Anyone out there have a functional DUNDi configuration using SIP for the inter-Asterisk transport? I've gotten it to work with IAX2, but if I change it to SIP it does not pass the call over even though it knows where to send it. Thanks. The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any.
I've trying to use DUNDi with SIP to see if it works around some limitations
of IAX2.
I do a DUNDi lookup, get my SIP path, and try to dial it. Asterisk immediately
says 'No such host', eventhough that's the path is just returned!
[Aug 2 13:07:05] == Spawn extension (global_vmdeposit, u9220371, 3) exited
non-zero on 'SIP/3254101-eb7d'
[Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", "***
OnNet originated call "Chocolate Chip" <3254101> ->
9220371") in new stack
[Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373",
"ipt/originator.py") in new stack
[Aug 2 13:07:13] -- Launched AGI Script
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug 2 13:07:13] -- AGI Script Executing Application: (SetAccount) Options:
(9220371)
[Aug 2 13:07:13] -- AGI Script Executing Application: (ChanIsAvail)
Options: (SIP/9220371)
[Aug 2 13:07:14] -- AGI Script Executing Application: (Dial) Options:
(SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371)
[Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 create_addr: No such host:
xxx.yyy.142.163/9220371
[Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)
[Aug 2 13:07:14] == Everyone is busy/congested at this time (1:0/0/1)
Not sure what is going on. I can see the query at the other end, but it
doesn't look like it ever receives the call.
Doug.
Secret? Do you mean sbsecret in sip.conf?> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, August 02, 2006 1:33 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] DUNDi with SIP > > > Using the SECRET variable for sip doesn't work. > > On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote: > > I've trying to use DUNDi with SIP to see if it works around > some limitations of IAX2. > > > > I do a DUNDi lookup, get my SIP path, and try to dial it. > Asterisk immediately says 'No such host', eventhough that's > the path is just returned! > > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, > u9220371, 3) exited non-zero on 'SIP/3254101-eb7d' > > [Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", > "*** OnNet originated call "Chocolate Chip" <3254101> -> > 9220371") in new stack > > [Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373", > "ipt/originator.py") in new stack > > [Aug 2 13:07:13] -- Launched AGI Script > /var/lib/asterisk/agi-bin/ipt/originator.py > > [Aug 2 13:07:13] -- AGI Script Executing Application: > (SetAccount) Options: (9220371) > > [Aug 2 13:07:13] -- AGI Script Executing Application: > (ChanIsAvail) Options: (SIP/9220371) > > [Aug 2 13:07:14] -- AGI Script Executing Application: > (Dial) Options: > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371) > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 > create_addr: No such host: xxx.yyy.142.163/9220371 > > [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040 > dial_exec_full: Unable to create channel of type 'SIP' (cause > 3 - No route to destination) > > [Aug 2 13:07:14] == Everyone is busy/congested at this > time (1:0/0/1) > > > > Not sure what is going on. I can see the query at the other > end, but it doesn't look like it ever receives the call. > > > > Doug. > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Aaron Daniel > Computer Systems Technician > Sam Houston State University > amdtech@shsu.edu > (936) 294-4198 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> -----Original Message----- > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > Sent: Wednesday, August 02, 2006 2:01 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: DUNDi with SIP > > > In article > <645FEC31A18FE54A8721500CDD55A7B6035D09C3@mail.oneeighty.com>, > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > I've trying to use DUNDi with SIP to see if it works around > some limitations of IAX2. > > > > I do a DUNDi lookup, get my SIP path, and try to dial it. > Asterisk immediately says 'No such > > host', eventhough that's the path is just returned! > > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, > u9220371, 3) exited non-zero on > > 'SIP/3254101-eb7d' > > [Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", > "*** OnNet originated call > > "Chocolate Chip" <3254101> -> 9220371") in new stack > > [Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373", > "ipt/originator.py") in new stack > > [Aug 2 13:07:13] -- Launched AGI Script > /var/lib/asterisk/agi-bin/ipt/originator.py > > [Aug 2 13:07:13] -- AGI Script Executing Application: > (SetAccount) Options: (9220371) > > [Aug 2 13:07:13] -- AGI Script Executing Application: > (ChanIsAvail) Options: (SIP/9220371) > > [Aug 2 13:07:14] -- AGI Script Executing Application: > (Dial) Options: > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371) > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 > create_addr: No such host: xxx.yyy.142.163/9220371 > > Try specifying the SIP argument as: > > SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@9220371@xxx.yyy.142.163 > > See the following line in the sample extensions.conf as an example: > > ;exten => > _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)Tony... it's DUNDi....
So what are the options?> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, August 02, 2006 2:03 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] DUNDi with SIP > > > I'm talking about the rotating DUNDi secret that is stored in dbsecret > in iax.conf. It doesn't exist in the SIP channel. > > On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote: > > Secret? Do you mean sbsecret in sip.conf? > > > > > -----Original Message----- > > > From: Aaron Daniel [mailto:amdtech@shsu.edu] > > > Sent: Wednesday, August 02, 2006 1:33 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [asterisk-users] DUNDi with SIP > > > > > > > > > Using the SECRET variable for sip doesn't work. > > > > > > On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote: > > > > I've trying to use DUNDi with SIP to see if it works around > > > some limitations of IAX2. > > > > > > > > I do a DUNDi lookup, get my SIP path, and try to dial it. > > > Asterisk immediately says 'No such host', eventhough that's > > > the path is just returned! > > > > > > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, > > > u9220371, 3) exited non-zero on 'SIP/3254101-eb7d' > > > > [Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", > > > "*** OnNet originated call "Chocolate Chip" <3254101> -> > > > 9220371") in new stack > > > > [Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373", > > > "ipt/originator.py") in new stack > > > > [Aug 2 13:07:13] -- Launched AGI Script > > > /var/lib/asterisk/agi-bin/ipt/originator.py > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > (SetAccount) Options: (9220371) > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > (ChanIsAvail) Options: (SIP/9220371) > > > > [Aug 2 13:07:14] -- AGI Script Executing Application: > > > (Dial) Options: > > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371) > > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 > > > create_addr: No such host: xxx.yyy.142.163/9220371 > > > > [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040 > > > dial_exec_full: Unable to create channel of type 'SIP' (cause > > > 3 - No route to destination) > > > > [Aug 2 13:07:14] == Everyone is busy/congested at this > > > time (1:0/0/1) > > > > > > > > Not sure what is going on. I can see the query at the other > > > end, but it doesn't look like it ever receives the call. > > > > > > > > Doug. > > > > > > > > > > > > _______________________________________________ > > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > > > asterisk-users mailing list > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > > > Aaron Daniel > > > Computer Systems Technician > > > Sam Houston State University > > > amdtech@shsu.edu > > > (936) 294-4198 > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > -- > Aaron Daniel > Computer Systems Technician > Sam Houston State University > amdtech@shsu.edu > (936) 294-4198 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
I've tried doing it without a username/password as described at:
http://www.voip-info.org/wiki/view/DUNDi+Enterprise+Configuration+SIP+with+no+passwords
but then authentication to the INVITE fails. I'm authenticating on the from:
field, ie the individual user, which I don't think is right.
I've also tried it with this in dundi.conf:
180netsip =>
global_dundi_local,1,SIP,dundisip:password@${IPADDR}/${NUMBER},nopartial
and this in sip.conf:
[dundisip]
type=user
context=global_dundi_local
secret=password
and I still get the 'create_addr: No such host: xxx.yyy.142.163/9220371'
messages on the client side.
Doug.
> -----Original Message-----
> From: Aaron Daniel [mailto:amdtech@shsu.edu]
> Sent: Wednesday, August 02, 2006 2:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] DUNDi with SIP
>
>
> I'm talking about the rotating DUNDi secret that is stored in dbsecret
> in iax.conf. It doesn't exist in the SIP channel.
>
> On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
> > Secret? Do you mean sbsecret in sip.conf?
> >
> > > -----Original Message-----
> > > From: Aaron Daniel [mailto:amdtech@shsu.edu]
> > > Sent: Wednesday, August 02, 2006 1:33 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] DUNDi with SIP
> > >
> > >
> > > Using the SECRET variable for sip doesn't work.
> > >
> > > On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
> > > > I've trying to use DUNDi with SIP to see if it works
around
> > > some limitations of IAX2.
> > > >
> > > > I do a DUNDi lookup, get my SIP path, and try to dial it.
> > > Asterisk immediately says 'No such host', eventhough
that's
> > > the path is just returned!
> > > >
> > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit,
> > > u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
> > > > [Aug 2 13:07:13] -- Executing
NoOp("SIP/3254101-6373",
> > > "*** OnNet originated call "Chocolate Chip"
<3254101> ->
> > > 9220371") in new stack
> > > > [Aug 2 13:07:13] -- Executing
AGI("SIP/3254101-6373",
> > > "ipt/originator.py") in new stack
> > > > [Aug 2 13:07:13] -- Launched AGI Script
> > > /var/lib/asterisk/agi-bin/ipt/originator.py
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (SetAccount) Options: (9220371)
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (ChanIsAvail) Options: (SIP/9220371)
> > > > [Aug 2 13:07:14] -- AGI Script Executing Application:
> > > (Dial) Options:
> > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371)
> > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980
> > > create_addr: No such host: xxx.yyy.142.163/9220371
> > > > [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040
> > > dial_exec_full: Unable to create channel of type 'SIP'
(cause
> > > 3 - No route to destination)
> > > > [Aug 2 13:07:14] == Everyone is busy/congested at this
> > > time (1:0/0/1)
> > > >
> > > > Not sure what is going on. I can see the query at the other
> > > end, but it doesn't look like it ever receives the call.
> > > >
> > > > Doug.
> > > >
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > --
> > > Aaron Daniel
> > > Computer Systems Technician
> > > Sam Houston State University
> > > amdtech@shsu.edu
> > > (936) 294-4198
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> amdtech@shsu.edu
> (936) 294-4198
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Alex,
Thanks... I haven't had any luck with it yet.
My dundi.conf has:
180netsip =>
global_dundi_local,1,SIP,dundisip:password@${IPADDR}/${NUMBER},nopartial
and my sip.conf has:
[dundisip]
type=user
context=global_dundi_local
secret=password
A DUNDI lookup on the console returns a SIP path:
*CLI> dundi lookup 9220370@180netsip
1. 1 SIP/dundisip:password@xxx.yyy.142.163/9220370 (EXISTS)
from 00:14:22:1e:2a:d0, expires in 0 s
DUNDi lookup completed in 129 ms
However, when I try to connect, I get a 'No such host' error...
*CLI> [Aug 2 14:18:43] -- Executing NoOp("SIP/3254101-a8d9",
"*** OnNet originated call "Chocolate Chip" <3254101> ->
9220371") in new stack
[Aug 2 14:18:43] -- Executing AGI("SIP/3254101-a8d9",
"ipt/originator.py") in new stack
[Aug 2 14:18:43] -- Launched AGI Script
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug 2 14:18:43] -- AGI Script Executing Application: (SetAccount) Options:
(9220371)
[Aug 2 14:18:43] -- AGI Script Executing Application: (ChanIsAvail)
Options: (SIP/9220371)
[Aug 2 14:18:43] -- AGI Script Executing Application: (Dial) Options:
(SIP/dundisip:password@xxx.yyy.142.163/9220371)
[Aug 2 14:18:43] WARNING[7842]: chan_sip.c:1980 create_addr: No such host:
xxx.yyy.142.163/9220371
[Aug 2 14:18:43] NOTICE[7842]: app_dial.c:1040 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)
[Aug 2 14:18:43] == Everyone is busy/congested at this time (1:0/0/1)
Doug.
-----Original Message-----
From: Alex Robar [mailto:alex.robar@gmail.com]
Sent: Wednesday, August 02, 2006 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi with SIP
You can use an unchanging password. It's not as secure, but it will provide
functionality.
Alex
On 8/2/06, Douglas Garstang < <mailto:dgarstang@oneeighty.com>
dgarstang@oneeighty.com> wrote:
So what are the options?
> -----Original Message-----
> From: Aaron Daniel [mailto: amdtech@shsu.edu]
> Sent: Wednesday, August 02, 2006 2:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] DUNDi with SIP
>
>
> I'm talking about the rotating DUNDi secret that is stored in dbsecret
> in iax.conf. It doesn't exist in the SIP channel.
>
> On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
> > Secret? Do you mean sbsecret in sip.conf?
> >
> > > -----Original Message-----
> > > From: Aaron Daniel [mailto: amdtech@shsu.edu]
> > > Sent: Wednesday, August 02, 2006 1:33 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] DUNDi with SIP
> > >
> > >
> > > Using the SECRET variable for sip doesn't work.
> > >
> > > On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
> > > > I've trying to use DUNDi with SIP to see if it works
around
> > > some limitations of IAX2.
> > > >
> > > > I do a DUNDi lookup, get my SIP path, and try to dial it.
> > > Asterisk immediately says 'No such host', eventhough
that's
> > > the path is just returned!
> > > >
> > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit,
> > > u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
> > > > [Aug 2 13:07:13] -- Executing
NoOp("SIP/3254101-6373",
> > > "*** OnNet originated call "Chocolate Chip"
<3254101> ->
> > > 9220371") in new stack
> > > > [Aug 2 13:07:13] -- Executing
AGI("SIP/3254101-6373",
> > > "ipt/originator.py") in new stack
> > > > [Aug 2 13:07:13] -- Launched AGI Script
> > > /var/lib/asterisk/agi-bin/ipt/originator.py
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (SetAccount) Options: (9220371)
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (ChanIsAvail) Options: (SIP/9220371)
> > > > [Aug 2 13:07:14] -- AGI Script Executing Application:
> > > (Dial) Options:
> > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371)
> > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980
> > > create_addr: No such host: xxx.yyy.142.163/9220371
> > > > [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040
> > > dial_exec_full: Unable to create channel of type 'SIP'
(cause
> > > 3 - No route to destination)
> > > > [Aug 2 13:07:14] == Everyone is busy/congested at this
> > > time (1:0/0/1)
> > > >
> > > > Not sure what is going on. I can see the query at the other
> > > end, but it doesn't look like it ever receives the call.
> > > >
> > > > Doug.
> > > >
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > --
> > > Aaron Daniel
> > > Computer Systems Technician
> > > Sam Houston State University
> > > amdtech@shsu.edu
> > > (936) 294-4198
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> --
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> amdtech@shsu.edu
> (936) 294-4198
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com
<http://Easynews.com> --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Alex Robar
alex.robar@gmail.com
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Alex,
Yep, I can dial 9220370 directly. I have two extensions on pbx1 and two on pbx2.
I can place calls from 9220371 to 9220370 which goes through pbx2 only, and all
is ok. 9220370 and 9220371 are registered on pbx2.
I had this all working with IAX. I didn't change the keys... so I would
assume that they would all still be ok. I haven't modified the keys or the
key definitions in dundi.conf.
Doug
-----Original Message-----
From: Alex Robar [mailto:alex.robar@gmail.com]
Sent: Wednesday, August 02, 2006 2:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi with SIP
Doug,
Two things: If you try to place that call manually (either via dialling it from
a phone that supports SP URIs or by making an ext. for it in your dialplan and
calling that extension), does it work properly? Are you able to place the call?
If not, is the CLI output the same as when you try it via DUNDi?
Second, are your keys generated properly, with public keys shared between the
two boxes OK? I had a lot of DUNDi problems initially, and found that my keys
were the problem.
Alex
On 8/2/06, Douglas Garstang < dgarstang@oneeighty.com> wrote:
Alex,
Thanks... I haven't had any luck with it yet.
My dundi.conf has:
180netsip =>
global_dundi_local,1,SIP,dundisip:password@${IPADDR}/${NUMBER},nopartial
and my sip.conf has:
[dundisip]
type=user
context=global_dundi_local
secret=password
A DUNDI lookup on the console returns a SIP path:
*CLI> dundi lookup 9220370@180netsip
1. 1 SIP/dundisip:password@xxx.yyy.142.163/9220370 (EXISTS)
from 00:14:22:1e:2a:d0, expires in 0 s
DUNDi lookup completed in 129 ms
However, when I try to connect, I get a 'No such host' error...
*CLI> [Aug 2 14:18:43] -- Executing NoOp("SIP/3254101-a8d9",
"*** OnNet originated call "Chocolate Chip" <3254101> ->
9220371") in new stack
[Aug 2 14:18:43] -- Executing AGI("SIP/3254101-a8d9",
"ipt/originator.py") in new stack
[Aug 2 14:18:43] -- Launched AGI Script
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug 2 14:18:43] -- AGI Script Executing Application: (SetAccount) Options:
(9220371)
[Aug 2 14:18:43] -- AGI Script Executing Application: (ChanIsAvail)
Options: (SIP/9220371)
[Aug 2 14:18:43] -- AGI Script Executing Application: (Dial) Options:
(SIP/dundisip:password@xxx.yyy.142.163/9220371)
[Aug 2 14:18:43] WARNING[7842]: chan_sip.c:1980 create_addr: No such host:
xxx.yyy.142.163/9220371
[Aug 2 14:18:43] NOTICE[7842]: app_dial.c:1040 dial_exec_full: Unable to create
channel of type 'SIP' (cause 3 - No route to destination)
[Aug 2 14:18:43] == Everyone is busy/congested at this time (1:0/0/1)
Doug.
-----Original Message-----
From: Alex Robar [mailto: alex.robar@gmail.com]
Sent: Wednesday, August 02, 2006 2:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DUNDi with SIP
You can use an unchanging password. It's not as secure, but it will provide
functionality.
Alex
On 8/2/06, Douglas Garstang < <mailto:dgarstang@oneeighty.com>
dgarstang@oneeighty.com> wrote:
So what are the options?
> -----Original Message-----
> From: Aaron Daniel [mailto: amdtech@shsu.edu]
> Sent: Wednesday, August 02, 2006 2:03 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] DUNDi with SIP
>
>
> I'm talking about the rotating DUNDi secret that is stored in dbsecret
> in iax.conf. It doesn't exist in the SIP channel.
>
> On Wed, 2006-08-02 at 13:43 -0600, Douglas Garstang wrote:
> > Secret? Do you mean sbsecret in sip.conf?
> >
> > > -----Original Message-----
> > > From: Aaron Daniel [mailto: amdtech@shsu.edu]
> > > Sent: Wednesday, August 02, 2006 1:33 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] DUNDi with SIP
> > >
> > >
> > > Using the SECRET variable for sip doesn't work.
> > >
> > > On Wed, 2006-08-02 at 13:11 -0600, Douglas Garstang wrote:
> > > > I've trying to use DUNDi with SIP to see if it works
around
> > > some limitations of IAX2.
> > > >
> > > > I do a DUNDi lookup, get my SIP path, and try to dial it.
> > > Asterisk immediately says 'No such host', eventhough
that's
> > > the path is just returned!
> > > >
> > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit,
> > > u9220371, 3) exited non-zero on 'SIP/3254101-eb7d'
> > > > [Aug 2 13:07:13] -- Executing
NoOp("SIP/3254101-6373",
> > > "*** OnNet originated call "Chocolate Chip"
<3254101> ->
> > > 9220371") in new stack
> > > > [Aug 2 13:07:13] -- Executing
AGI("SIP/3254101-6373",
> > > "ipt/originator.py") in new stack
> > > > [Aug 2 13:07:13] -- Launched AGI Script
> > > /var/lib/asterisk/agi-bin/ipt/originator.py
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (SetAccount) Options: (9220371)
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (ChanIsAvail) Options: (SIP/9220371)
> > > > [Aug 2 13:07:14] -- AGI Script Executing Application:
> > > (Dial) Options:
> > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371)
> > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980
> > > create_addr: No such host: xxx.yyy.142.163/9220371
> > > > [Aug 2 13:07:14] NOTICE[5429]: app_dial.c:1040
> > > dial_exec_full: Unable to create channel of type 'SIP'
(cause
> > > 3 - No route to destination)
> > > > [Aug 2 13:07:14] == Everyone is busy/congested at this
> > > time (1:0/0/1)
> > > >
> > > > Not sure what is going on. I can see the query at the other
> > > end, but it doesn't look like it ever receives the call.
> > > >
> > > > Doug.
> > > >
> > > >
> > > > _______________________________________________
> > > > --Bandwidth and Colocation provided by Easynews.com --
> > > >
> > > > asterisk-users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > --
> > > Aaron Daniel
> > > Computer Systems Technician
> > > Sam Houston State University
> > > amdtech@shsu.edu
> > > (936) 294-4198
> > > _______________________________________________
> > > --Bandwidth and Colocation provided by Easynews.com --
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
> > >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
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> > http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
> --
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> amdtech@shsu.edu
> (936) 294-4198
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com
<http://Easynews.com> --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
<http://lists.digium.com/mailman/listinfo/asterisk-users>
>
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Alex Robar
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> -----Original Message----- > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > Sent: Wednesday, August 02, 2006 3:49 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: DUNDi with SIP > > > In article > <645FEC31A18FE54A8721500CDD55A7B6035D09C5@mail.oneeighty.com>, > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > -----Original Message----- > > > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > > > Sent: Wednesday, August 02, 2006 2:01 PM > > > To: asterisk-users@lists.digium.com > > > Subject: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > In article > > > <645FEC31A18FE54A8721500CDD55A7B6035D09C3@mail.oneeighty.com>, > > > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > > I've trying to use DUNDi with SIP to see if it works around > > > some limitations of IAX2. > > > > > > > > I do a DUNDi lookup, get my SIP path, and try to dial it. > > > Asterisk immediately says 'No such > > > > host', eventhough that's the path is just returned! > > > > > > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, > > > u9220371, 3) exited non-zero on > > > > 'SIP/3254101-eb7d' > > > > [Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", > > > "*** OnNet originated call > > > > "Chocolate Chip" <3254101> -> 9220371") in new stack > > > > [Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373", > > > "ipt/originator.py") in new stack > > > > [Aug 2 13:07:13] -- Launched AGI Script > > > /var/lib/asterisk/agi-bin/ipt/originator.py > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > (SetAccount) Options: (9220371) > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > (ChanIsAvail) Options: (SIP/9220371) > > > > [Aug 2 13:07:14] -- AGI Script Executing Application: > > > (Dial) Options: > > > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371) > > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 > > > create_addr: No such host: xxx.yyy.142.163/9220371 > > > > > > Try specifying the SIP argument as: > > > > > > SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@9220371@xxx.yyy.142.163 > > > > > > See the following line in the sample extensions.conf as > an example: > > > > > > ;exten => > > > _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) > > > > Tony... it's DUNDi.... > > OK, I know nothing about DUNDi. I was only highlighting what appeared > to be invalid or at least ambiguous syntax in the SIP channel > requested. > SIP appears not to like SIP/user:pass@host/number, but instead wants > SIP/user:pass@number@host > > Unless you can set up a sip.conf friend entry [xxxx] and then use > SIP/xxxx/number > > Hope this helps. If not, oh well.Well yes, it looked dubious to me too, although I can't find the syntaxt documented anywhere. However, that's what DUNDis giving me as a path to the phone! Something is screwed with DUNDi and SIP. Has ANYONE actually implemnted it? I can't find it documented anywhere Doug.
> -----Original Message----- > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > Sent: Wednesday, August 02, 2006 3:49 PM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Re: DUNDi with SIP > > > In article > <645FEC31A18FE54A8721500CDD55A7B6035D09C5@mail.oneeighty.com>, > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > -----Original Message----- > > > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > > > Sent: Wednesday, August 02, 2006 2:01 PM > > > To: asterisk-users@lists.digium.com > > > Subject: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > In article > > > <645FEC31A18FE54A8721500CDD55A7B6035D09C3@mail.oneeighty.com>, > > > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > > I've trying to use DUNDi with SIP to see if it works around > > > some limitations of IAX2. > > > > > > > > I do a DUNDi lookup, get my SIP path, and try to dial it. > > > Asterisk immediately says 'No such > > > > host', eventhough that's the path is just returned! > > > > > > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, > > > u9220371, 3) exited non-zero on > > > > 'SIP/3254101-eb7d' > > > > [Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", > > > "*** OnNet originated call > > > > "Chocolate Chip" <3254101> -> 9220371") in new stack > > > > [Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373", > > > "ipt/originator.py") in new stack > > > > [Aug 2 13:07:13] -- Launched AGI Script > > > /var/lib/asterisk/agi-bin/ipt/originator.py > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > (SetAccount) Options: (9220371) > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > (ChanIsAvail) Options: (SIP/9220371) > > > > [Aug 2 13:07:14] -- AGI Script Executing Application: > > > (Dial) Options: > > > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371) > > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 > > > create_addr: No such host: xxx.yyy.142.163/9220371 > > > > > > Try specifying the SIP argument as: > > > > > > SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@9220371@xxx.yyy.142.163 > > > > > > See the following line in the sample extensions.conf as > an example: > > > > > > ;exten => > > > _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) > > > > Tony... it's DUNDi.... > > OK, I know nothing about DUNDi. I was only highlighting what appeared > to be invalid or at least ambiguous syntax in the SIP channel > requested. > SIP appears not to like SIP/user:pass@host/number, but instead wants > SIP/user:pass@number@host > > Unless you can set up a sip.conf friend entry [xxxx] and then use > SIP/xxxx/number > > Hope this helps. If not, oh well.Tony, I was able to fiddle with dundi.conf, and am now getting a SIP path in the format SIP/user:password@number@host: *CLI> dundi lookup 9220370@180netsip 1. 1 SIP/dundisip:password@9220370@xxx.yyy.142.163 (EXISTS) from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in 171 ms However, when I try to dial this, I am still getting: [Aug 2 15:57:35] WARNING[9916]: chan_sip.c:1980 create_addr: No such host: 9220370@xxx.yyy.142.163 [Aug 2 15:57:35] NOTICE[9916]: app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Arrgh
The way to make this work is to define a sip user/peer with the IP
address in it, then have your dundi.conf entry look like:
180netsip => global_dundi_local,1,SIP/peername/${NUMBER},nopartial
As far as I can tell from the code, this is the only way to make it work
properly based on the way the string sent to the channel driver is being
parsed.
- Brad
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Wednesday, August 02, 2006 5:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
> -----Original Message-----
> From: Tony Mountifield [mailto:tony@softins.clara.co.uk]
> Sent: Wednesday, August 02, 2006 3:49 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Re: DUNDi with SIP
>
>
> In article
> <645FEC31A18FE54A8721500CDD55A7B6035D09C5@mail.oneeighty.com>,
> Douglas Garstang <dgarstang@oneeighty.com> wrote:
> > > -----Original Message-----
> > > From: Tony Mountifield [mailto:tony@softins.clara.co.uk]
> > > Sent: Wednesday, August 02, 2006 2:01 PM
> > > To: asterisk-users@lists.digium.com
> > > Subject: [asterisk-users] Re: DUNDi with SIP
> > >
> > >
> > > In article
> > >
<645FEC31A18FE54A8721500CDD55A7B6035D09C3@mail.oneeighty.com>,
> > > Douglas Garstang <dgarstang@oneeighty.com> wrote:
> > > > I've trying to use DUNDi with SIP to see if it works
around
> > > some limitations of IAX2.
> > > >
> > > > I do a DUNDi lookup, get my SIP path, and try to dial it.
> > > Asterisk immediately says 'No such
> > > > host', eventhough that's the path is just returned!
> > > >
> > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit,
> > > u9220371, 3) exited non-zero on
> > > > 'SIP/3254101-eb7d'
> > > > [Aug 2 13:07:13] -- Executing
NoOp("SIP/3254101-6373",
> > > "*** OnNet originated call
> > > > "Chocolate Chip" <3254101> ->
9220371") in new stack
> > > > [Aug 2 13:07:13] -- Executing
AGI("SIP/3254101-6373",
> > > "ipt/originator.py") in new stack
> > > > [Aug 2 13:07:13] -- Launched AGI Script
> > > /var/lib/asterisk/agi-bin/ipt/originator.py
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (SetAccount) Options: (9220371)
> > > > [Aug 2 13:07:13] -- AGI Script Executing Application:
> > > (ChanIsAvail) Options: (SIP/9220371)
> > > > [Aug 2 13:07:14] -- AGI Script Executing Application:
> > > (Dial) Options:
> > > >
(SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371)
> > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980
> > > create_addr: No such host: xxx.yyy.142.163/9220371
> > >
> > > Try specifying the SIP argument as:
> > >
> > > SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@9220371@xxx.yyy.142.163
> > >
> > > See the following line in the sample extensions.conf as
> an example:
> > >
> > > ;exten =>
> > > _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)
> >
> > Tony... it's DUNDi....
>
> OK, I know nothing about DUNDi. I was only highlighting what appeared
> to be invalid or at least ambiguous syntax in the SIP channel
> requested.
> SIP appears not to like SIP/user:pass@host/number, but instead wants
> SIP/user:pass@number@host
>
> Unless you can set up a sip.conf friend entry [xxxx] and then use
> SIP/xxxx/number
>
> Hope this helps. If not, oh well.
Tony, I was able to fiddle with dundi.conf, and am now getting a SIP
path in the format SIP/user:password@number@host:
*CLI> dundi lookup 9220370@180netsip
1. 1 SIP/dundisip:password@9220370@xxx.yyy.142.163 (EXISTS)
from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in
171 ms
However, when I try to dial this, I am still getting:
[Aug 2 15:57:35] WARNING[9916]: chan_sip.c:1980 create_addr: No such
host: 9220370@xxx.yyy.142.163 [Aug 2 15:57:35] NOTICE[9916]:
app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP'
(cause 3 - No route to destination)
Arrgh
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So... given then that there's multiple systems in the DUNDI domain (obviously), that means there's going to be multiple hosts, and therefore I'll need to have one sip user for each host, and one dundi entry for each user as well... Bleck.> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Wednesday, August 02, 2006 4:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > The way to make this work is to define a sip user/peer with the IP > address in it, then have your dundi.conf entry look like: > > 180netsip => global_dundi_local,1,SIP/peername/${NUMBER},nopartial > > As far as I can tell from the code, this is the only way to > make it work > properly based on the way the string sent to the channel > driver is being > parsed. > > - Brad > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas > Garstang > Sent: Wednesday, August 02, 2006 5:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > -----Original Message----- > > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > > Sent: Wednesday, August 02, 2006 3:49 PM > > To: asterisk-users@lists.digium.com > > Subject: [asterisk-users] Re: DUNDi with SIP > > > > > > In article > > <645FEC31A18FE54A8721500CDD55A7B6035D09C5@mail.oneeighty.com>, > > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > > -----Original Message----- > > > > From: Tony Mountifield [mailto:tony@softins.clara.co.uk] > > > > Sent: Wednesday, August 02, 2006 2:01 PM > > > > To: asterisk-users@lists.digium.com > > > > Subject: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > > > > In article > > > > <645FEC31A18FE54A8721500CDD55A7B6035D09C3@mail.oneeighty.com>, > > > > Douglas Garstang <dgarstang@oneeighty.com> wrote: > > > > > I've trying to use DUNDi with SIP to see if it works around > > > > some limitations of IAX2. > > > > > > > > > > I do a DUNDi lookup, get my SIP path, and try to dial it. > > > > Asterisk immediately says 'No such > > > > > host', eventhough that's the path is just returned! > > > > > > > > > > [Aug 2 13:07:05] == Spawn extension (global_vmdeposit, > > > > u9220371, 3) exited non-zero on > > > > > 'SIP/3254101-eb7d' > > > > > [Aug 2 13:07:13] -- Executing NoOp("SIP/3254101-6373", > > > > "*** OnNet originated call > > > > > "Chocolate Chip" <3254101> -> 9220371") in new stack > > > > > [Aug 2 13:07:13] -- Executing AGI("SIP/3254101-6373", > > > > "ipt/originator.py") in new stack > > > > > [Aug 2 13:07:13] -- Launched AGI Script > > > > /var/lib/asterisk/agi-bin/ipt/originator.py > > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > > (SetAccount) Options: (9220371) > > > > > [Aug 2 13:07:13] -- AGI Script Executing Application: > > > > (ChanIsAvail) Options: (SIP/9220371) > > > > > [Aug 2 13:07:14] -- AGI Script Executing Application: > > > > (Dial) Options: > > > > > (SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@xxx.yyy.142.163/9220371) > > > > > [Aug 2 13:07:14] WARNING[5429]: chan_sip.c:1980 > > > > create_addr: No such host: xxx.yyy.142.163/9220371 > > > > > > > > Try specifying the SIP argument as: > > > > > > > > SIP/dundisip:dU7wMVIkX00hbS5OFaZBDQ@9220371@xxx.yyy.142.163 > > > > > > > > See the following line in the sample extensions.conf as > > an example: > > > > > > > > ;exten => > > > > _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT) > > > > > > Tony... it's DUNDi.... > > > > OK, I know nothing about DUNDi. I was only highlighting > what appeared > > to be invalid or at least ambiguous syntax in the SIP channel > > requested. > > SIP appears not to like SIP/user:pass@host/number, but > instead wants > > SIP/user:pass@number@host > > > > Unless you can set up a sip.conf friend entry [xxxx] and then use > > SIP/xxxx/number > > > > Hope this helps. If not, oh well. > > Tony, I was able to fiddle with dundi.conf, and am now getting a SIP > path in the format SIP/user:password@number@host: > > *CLI> dundi lookup 9220370@180netsip > 1. 1 SIP/dundisip:password@9220370@xxx.yyy.142.163 (EXISTS) > from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in > 171 ms > > However, when I try to dial this, I am still getting: > > [Aug 2 15:57:35] WARNING[9916]: chan_sip.c:1980 create_addr: No such > host: 9220370@xxx.yyy.142.163 [Aug 2 15:57:35] NOTICE[9916]: > app_dial.c:1040 dial_exec_full: Unable to create channel of type 'SIP' > (cause 3 - No route to destination) > > Arrgh > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > The contents of this e-mail are intended for the named > addressee only. It contains information that may be > confidential. Unless you are the named addressee or an > authorized designee, you may not copy or use it, or disclose > it to anyone else. If you received it in error please notify > us immediately and then destroy it. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Wednesday, August 02, 2006 4:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > The way to make this work is to define a sip user/peer with the IP > address in it, then have your dundi.conf entry look like: > > 180netsip => global_dundi_local,1,SIP/peername/${NUMBER},nopartial > > As far as I can tell from the code, this is the only way to > make it work > properly based on the way the string sent to the channel > driver is being > parsed.You this, this is: a) completely different to what's in dundi.conf right now b) completely different to the docs posted on the wiki. Have people actually TRIED this, or do they just what they _THINK_ will work as docs??? Doug.
> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Wednesday, August 02, 2006 4:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > The way to make this work is to define a sip user/peer with the IP > address in it, then have your dundi.conf entry look like: > > 180netsip => global_dundi_local,1,SIP/peername/${NUMBER},nopartial > > As far as I can tell from the code, this is the only way to > make it work > properly based on the way the string sent to the channel > driver is being > parsed.No luck Brad. dundi.conf has as you suggested: 180netsip => global_dundi_local,100,SIP/dundi_pbx1/${NUMBER},nopartial 180netsip => global_dundi_local,200,SIP/dundi_pbx2/${NUMBER},nopartial and dundi.conf has: [dundi_pbx1] type=peer context=global_dundi_local host=labpbx1.ipt.oneeighty.com [dundi_pbx2] type=peer context=global_dundi_local host=labpbx2.ipt.oneeighty.com and when I do a query, I get: dundi lookup 9220371@180netsip 1. 200 Unknown/nopartial (EXISTS|CANMATCH) from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in 121 ms Obviously, something is seriously screwed up. Doug.
> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, August 02, 2006 4:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote: > > Well yes, it looked dubious to me too, although I can't > find the syntaxt documented anywhere. > > However, that's what DUNDis giving me as a path to the phone! > > > > Something is screwed with DUNDi and SIP. Has ANYONE > actually implemnted it? > > I can't find it documented anywhere > > > > Doug. > > DUNDi gives you only what you give it to give you. You're > the one that > needs to set the dial string correctly in DUNDi to get one back that > works. DUNDi is only as automatic as you let it be. > > This is what ours looks like. We don't use the iax versions (mainly > cause I want a homogenous SIP system), but we have entries in sip.conf > include files for each of the servers so we just dial > ${server}/${number}. This has been working for us for about 2 months > now, pretty much flawlessly as long as the phone's registered. > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} > e164-iax => dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > internal => dundi-extens,0,SIP,scm1/${NUMBER} > internal-iax => dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > [scm1] > type=friend > secret=p4ssw0rd > insecure=very > context=incoming > host=scm1.shsu.edu > qualify=yes > nat=noThanks Aaron, but I don't understand how that can work. Don't you have more than one host in your DUNDi domain? Doug.
Errr... I should learn to pay attention to what I'm writing...
Actually, try:
180netsip => global_dundi_local,100,SIP,dundi_pbx1/${NUMBER},nopartial
I was thinking in terms of Dial() arguments when I wrote the
SIP/peername/number instead of SIP,peername/number syntax.
My bad...
- Brad
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Wednesday, August 02, 2006 6:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
> -----Original Message-----
> From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
> Sent: Wednesday, August 02, 2006 4:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> The way to make this work is to define a sip user/peer with the IP
> address in it, then have your dundi.conf entry look like:
>
> 180netsip => global_dundi_local,1,SIP/peername/${NUMBER},nopartial
>
> As far as I can tell from the code, this is the only way to make it
> work properly based on the way the string sent to the channel driver
> is being parsed.
No luck Brad.
dundi.conf has as you suggested:
180netsip => global_dundi_local,100,SIP/dundi_pbx1/${NUMBER},nopartial
180netsip => global_dundi_local,200,SIP/dundi_pbx2/${NUMBER},nopartial
and dundi.conf has:
[dundi_pbx1]
type=peer
context=global_dundi_local
host=labpbx1.ipt.oneeighty.com
[dundi_pbx2]
type=peer
context=global_dundi_local
host=labpbx2.ipt.oneeighty.com
and when I do a query, I get:
dundi lookup 9220371@180netsip
1. 200 Unknown/nopartial (EXISTS|CANMATCH)
from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in
121 ms
Obviously, something is seriously screwed up.
Doug.
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> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, August 02, 2006 4:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote: > > Well yes, it looked dubious to me too, although I can't > find the syntaxt documented anywhere. > > However, that's what DUNDis giving me as a path to the phone! > > > > Something is screwed with DUNDi and SIP. Has ANYONE > actually implemnted it? > > I can't find it documented anywhere > > > > Doug. > > DUNDi gives you only what you give it to give you. You're > the one that > needs to set the dial string correctly in DUNDi to get one back that > works. DUNDi is only as automatic as you let it be. > > This is what ours looks like. We don't use the iax versions (mainly > cause I want a homogenous SIP system), but we have entries in sip.conf > include files for each of the servers so we just dial > ${server}/${number}. This has been working for us for about 2 months > now, pretty much flawlessly as long as the phone's registered. > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} > e164-iax => dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > internal => dundi-extens,0,SIP,scm1/${NUMBER} > internal-iax => dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > [scm1] > type=friend > secret=p4ssw0rd > insecure=very > context=incoming > host=scm1.shsu.edu > qualify=yes > nat=noAaron, while not really sure what I was doing, but extending beyond your example, I gave this a shot: dundi.conf: 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER} 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER} sip.conf: [dundisip1] type=friend secret=password insecure=very context=global_dundi_local host=labpbx1.ipt.twoeighty.com qualify=yes [dundisip2] type=friend secret=password insecure=very context=global_dundi_local host=labpbx2.ipt.twoeighty.com qualify=yes A CLI lookup looks better... *CLI> dundi lookup 9220370@180netsip 1. 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH) from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in 107 ms However, my dial still fails... [Aug 2 16:42:44] -- AGI Script Executing Application: (Dial) Options: (SIP/dundisip2/9220370) [Aug 2 16:42:44] -- Called dundisip2/9220370 [Aug 2 16:42:44] NOTICE[10474]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '"Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1' [Aug 2 16:42:44] -- SIP/dundisip2-7b30 is circuit-busy
> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Wednesday, August 02, 2006 4:45 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > Errr... I should learn to pay attention to what I'm writing... > > Actually, try: > > 180netsip => global_dundi_local,100,SIP,dundi_pbx1/${NUMBER},nopartial > > I was thinking in terms of Dial() arguments when I wrote the > SIP/peername/number instead of SIP,peername/number syntax. > > My bad...Brad, that's pretty much what Aaron just posted with. Thanks. I've made a subequent post trying what you/he suggested but I'm still getting some errors. Doug.
Try putting a username= in the peer (BTW, use peer not friend) definitions. You appear to be attempting to authenticate as the originating callerid (3254101). - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang Sent: Wednesday, August 02, 2006 6:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: DUNDi with SIP> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, August 02, 2006 4:02 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote: > > Well yes, it looked dubious to me too, although I can't > find the syntaxt documented anywhere. > > However, that's what DUNDis giving me as a path to the phone! > > > > Something is screwed with DUNDi and SIP. Has ANYONE > actually implemnted it? > > I can't find it documented anywhere > > > > Doug. > > DUNDi gives you only what you give it to give you. You're the one > that needs to set the dial string correctly in DUNDi to get one back > that works. DUNDi is only as automatic as you let it be. > > This is what ours looks like. We don't use the iax versions (mainly > cause I want a homogenous SIP system), but we have entries in sip.conf> include files for each of the servers so we just dial > ${server}/${number}. This has been working for us for about 2 months > now, pretty much flawlessly as long as the phone's registered. > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax => > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax => > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > [scm1] > type=friend > secret=p4ssw0rd > insecure=very > context=incoming > host=scm1.shsu.edu > qualify=yes > nat=noAaron, while not really sure what I was doing, but extending beyond your example, I gave this a shot: dundi.conf: 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER} 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER} sip.conf: [dundisip1] type=friend secret=password insecure=very context=global_dundi_local host=labpbx1.ipt.twoeighty.com qualify=yes [dundisip2] type=friend secret=password insecure=very context=global_dundi_local host=labpbx2.ipt.twoeighty.com qualify=yes A CLI lookup looks better... *CLI> dundi lookup 9220370@180netsip 1. 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH) from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in 107 ms However, my dial still fails... [Aug 2 16:42:44] -- AGI Script Executing Application: (Dial) Options: (SIP/dundisip2/9220370) [Aug 2 16:42:44] -- Called dundisip2/9220370 [Aug 2 16:42:44] NOTICE[10474]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '"Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1' [Aug 2 16:42:44] -- SIP/dundisip2-7b30 is circuit-busy _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.
Ok, but don't you need to have [scm1] AND [sgw1] in sip.conf?> -----Original Message----- > From: Aaron Daniel [mailto:amdtech@shsu.edu] > Sent: Wednesday, August 02, 2006 4:56 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > You don't have to have every host in your dundi.conf files. The way > we've got ours set up, each server tells the other servers > how to reach > it. For example, our primary call server (scm1) publishes > what numbers > it can handle, so I only list it's contexts in it's own file. Then, > sgw1 publishes what IT can handle. There's a matching e164 > and internal > context on each server to tell the others what it can take. > > On Wed, 2006-08-02 at 16:35 -0600, Douglas Garstang wrote: > > > -----Original Message----- > > > From: Aaron Daniel [mailto:amdtech@shsu.edu] > > > Sent: Wednesday, August 02, 2006 4:02 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote: > > > > Well yes, it looked dubious to me too, although I can't > > > find the syntaxt documented anywhere. > > > > However, that's what DUNDis giving me as a path to the phone! > > > > > > > > Something is screwed with DUNDi and SIP. Has ANYONE > > > actually implemnted it? > > > > I can't find it documented anywhere > > > > > > > > Doug. > > > > > > DUNDi gives you only what you give it to give you. You're > > > the one that > > > needs to set the dial string correctly in DUNDi to get > one back that > > > works. DUNDi is only as automatic as you let it be. > > > > > > This is what ours looks like. We don't use the iax > versions (mainly > > > cause I want a homogenous SIP system), but we have > entries in sip.conf > > > include files for each of the servers so we just dial > > > ${server}/${number}. This has been working for us for > about 2 months > > > now, pretty much flawlessly as long as the phone's registered. > > > > > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} > > > e164-iax => dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > > internal => dundi-extens,0,SIP,scm1/${NUMBER} > > > internal-iax => dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > > > > > [scm1] > > > type=friend > > > secret=p4ssw0rd > > > insecure=very > > > context=incoming > > > host=scm1.shsu.edu > > > qualify=yes > > > nat=no > > > > Thanks Aaron, but I don't understand how that can work. > Don't you have more than one host in your DUNDi domain? > > > > Doug. > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Bradley,
I changed the type from friend to peer in sip.conf...
[dundisip1]
type=peer
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.twoeighty.com
qualify=yes
[dundisip2]
type=peer
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.twoeighty.com
qualify=yes
but that just yielded the same error...
[Aug 2 17:07:51] NOTICE[10971]: chan_sip.c:9685 handle_response_invite: Failed
to authenticate on INVITE to '"Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as5e6e3efe'
Btw, here's a sip trace between the asterisk boxes...
Capturing on eth0
1 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
2 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 407 Proxy
Authentication Required
3 0.177497 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@labpbx1.ipt.oneeighty.com
4 0.329036 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
5 0.329492 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
6 0.643513 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP Request: INVITE
sip:9220370@labpbx2.ipt.twoeighty.com, with session description
7 0.644104 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: 407 Proxy
Authentication Required
8 0.644356 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: ACK
sip:9220370@labpbx2.ipt.oneeighty.com
9 0.647990 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP Status: 200 OK, with
session description
10 0.806136 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@xxx.yyy.142.162
11 2.280664 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: BYE
sip:9220370@xxx.yyy.142.162
12 2.280780 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 200 OK
You can see that the first system doesn't resend the INVITE with the auth
credentials as requested.
So, then I also put the username field, and sip.conf now looks like this:
[dundisip1]
type=peer
username=dundisip1
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.twoeighty.com
qualify=yes
[dundisip2]
type=peer
username=dundisip2
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.twoeeighty.com
qualify=yes
and with dundi.conf as:
180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
A CLI lookup still yields:
*CLI> dundi lookup 9220370@180netsip
1. 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
from 00:14:22:1e:2a:d0, expires in 0 s
DUNDi lookup completed in 130 ms
HOWEVER, at attempt to dial results in this now:
[Aug 2 17:14:00] WARNING[11178]: chan_sip.c:9696 handle_response_invite:
Forbidden - wrong password on authentication for INVITE to '"Chocolate
Chip" <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f'
And here's the SIP trace for THAT!
1 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
2 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 407 Proxy
Authentication Required
3 0.158008 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@labpbx1.ipt.twoeighty.com
4 0.295054 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
5 0.300557 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
6 0.602324 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP Request: INVITE
sip:9220370@labpbx2.ipt.twoeighty.com, with session description
7 0.603002 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: 407 Proxy
Authentication Required
8 0.603303 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: ACK
sip:9220370@labpbx2.ipt.twoeighty.com
9 0.603485 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP Request: INVITE
sip:9220370@labpbx2.ipt.twoeighty.com, with session description
10 0.604251 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: 403 Forbidden
11 0.604553 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: ACK
sip:9220370@labpbx2.ipt.twoeighty.com
12 0.608324 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP Status: 200 OK, with
session description
13 0.749668 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@xxx.yyy.142.162
And on the second Asterisk console we have logged:
Aug 2 17:13:57 NOTICE[29764]: chan_sip.c:10469 handle_request_invite: Failed to
authenticate user "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as6090e60f
Ugh... not having much luck with this...
> -----Original Message-----
> From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
> Sent: Wednesday, August 02, 2006 5:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> Try putting a username= in the peer (BTW, use peer not friend)
> definitions. You appear to be attempting to authenticate as the
> originating callerid (3254101).
>
> - Brad
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
> Garstang
> Sent: Wednesday, August 02, 2006 6:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
> > -----Original Message-----
> > From: Aaron Daniel [mailto:amdtech@shsu.edu]
> > Sent: Wednesday, August 02, 2006 4:02 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> >
> >
> > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
> > > Well yes, it looked dubious to me too, although I can't
> > find the syntaxt documented anywhere.
> > > However, that's what DUNDis giving me as a path to the phone!
> > >
> > > Something is screwed with DUNDi and SIP. Has ANYONE
> > actually implemnted it?
> > > I can't find it documented anywhere
> > >
> > > Doug.
> >
> > DUNDi gives you only what you give it to give you. You're the one
> > that needs to set the dial string correctly in DUNDi to get
> one back
> > that works. DUNDi is only as automatic as you let it be.
> >
> > This is what ours looks like. We don't use the iax
> versions (mainly
> > cause I want a homogenous SIP system), but we have entries
> in sip.conf
>
> > include files for each of the servers so we just dial
> > ${server}/${number}. This has been working for us for
> about 2 months
> > now, pretty much flawlessly as long as the phone's registered.
> >
> > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax =>
> > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax =>
> > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> >
> > [scm1]
> > type=friend
> > secret=p4ssw0rd
> > insecure=very
> > context=incoming
> > host=scm1.shsu.edu
> > qualify=yes
> > nat=no
>
> Aaron, while not really sure what I was doing, but extending
> beyond your
> example, I gave this a shot:
>
> dundi.conf:
> 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
> 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
>
> sip.conf:
> [dundisip1]
> type=friend
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx1.ipt.twoeighty.com
> qualify=yes
>
> [dundisip2]
> type=friend
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx2.ipt.twoeighty.com
> qualify=yes
>
> A CLI lookup looks better...
>
> *CLI> dundi lookup 9220370@180netsip
> 1. 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
> from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in
> 107 ms
>
> However, my dial still fails...
>
> [Aug 2 16:42:44] -- AGI Script Executing Application: (Dial)
> Options: (SIP/dundisip2/9220370)
> [Aug 2 16:42:44] -- Called dundisip2/9220370
> [Aug 2 16:42:44] NOTICE[10474]: chan_sip.c:9685
> handle_response_invite:
> Failed to authenticate on INVITE to '"Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1'
> [Aug 2 16:42:44] -- SIP/dundisip2-7b30 is circuit-busy
>
>
> _______________________________________________
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>
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> The contents of this e-mail are intended for the named
> addressee only. It contains information that may be
> confidential. Unless you are the named addressee or an
> authorized designee, you may not copy or use it, or disclose
> it to anyone else. If you received it in error please notify
> us immediately and then destroy it.
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Could you perhaps post a sip debug from both of the Asterisk consoles for the
respective peers?
I think we're very close, and I definitely want to get this working for you.
Regards,
- Brad
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Douglas Garstang
Sent: Wed 8/2/2006 7:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Bradley,
I changed the type from friend to peer in sip.conf...
[dundisip1]
type=peer
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.twoeighty.com
qualify=yes
[dundisip2]
type=peer
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.twoeighty.com
qualify=yes
but that just yielded the same error...
[Aug 2 17:07:51] NOTICE[10971]: chan_sip.c:9685 handle_response_invite: Failed
to authenticate on INVITE to '"Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as5e6e3efe'
Btw, here's a sip trace between the asterisk boxes...
Capturing on eth0
1 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
2 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 407 Proxy
Authentication Required
3 0.177497 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@labpbx1.ipt.oneeighty.com
4 0.329036 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
5 0.329492 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
6 0.643513 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP Request: INVITE
sip:9220370@labpbx2.ipt.twoeighty.com, with session description
7 0.644104 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: 407 Proxy
Authentication Required
8 0.644356 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: ACK
sip:9220370@labpbx2.ipt.oneeighty.com
9 0.647990 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP Status: 200 OK, with
session description
10 0.806136 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@xxx.yyy.142.162
11 2.280664 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: BYE
sip:9220370@xxx.yyy.142.162
12 2.280780 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 200 OK
You can see that the first system doesn't resend the INVITE with the auth
credentials as requested.
So, then I also put the username field, and sip.conf now looks like this:
[dundisip1]
type=peer
username=dundisip1
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.twoeighty.com
qualify=yes
[dundisip2]
type=peer
username=dundisip2
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.twoeeighty.com
qualify=yes
and with dundi.conf as:
180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
A CLI lookup still yields:
*CLI> dundi lookup 9220370@180netsip
1. 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
from 00:14:22:1e:2a:d0, expires in 0 s
DUNDi lookup completed in 130 ms
HOWEVER, at attempt to dial results in this now:
[Aug 2 17:14:00] WARNING[11178]: chan_sip.c:9696 handle_response_invite:
Forbidden - wrong password on authentication for INVITE to '"Chocolate
Chip" <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f'
And here's the SIP trace for THAT!
1 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
2 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 407 Proxy
Authentication Required
3 0.158008 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@labpbx1.ipt.twoeighty.com
4 0.295054 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP Request: INVITE
sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with session description
5 0.300557 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
6 0.602324 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP Request: INVITE
sip:9220370@labpbx2.ipt.twoeighty.com, with session description
7 0.603002 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: 407 Proxy
Authentication Required
8 0.603303 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: ACK
sip:9220370@labpbx2.ipt.twoeighty.com
9 0.603485 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP Request: INVITE
sip:9220370@labpbx2.ipt.twoeighty.com, with session description
10 0.604251 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: 403 Forbidden
11 0.604553 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: ACK
sip:9220370@labpbx2.ipt.twoeighty.com
12 0.608324 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP Status: 200 OK, with
session description
13 0.749668 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: ACK
sip:9220370@xxx.yyy.142.162
And on the second Asterisk console we have logged:
Aug 2 17:13:57 NOTICE[29764]: chan_sip.c:10469 handle_request_invite: Failed to
authenticate user "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as6090e60f
Ugh... not having much luck with this...
> -----Original Message-----
> From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
> Sent: Wednesday, August 02, 2006 5:01 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> Try putting a username= in the peer (BTW, use peer not friend)
> definitions. You appear to be attempting to authenticate as the
> originating callerid (3254101).
>
> - Brad
>
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
> Garstang
> Sent: Wednesday, August 02, 2006 6:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
> > -----Original Message-----
> > From: Aaron Daniel [mailto:amdtech@shsu.edu]
> > Sent: Wednesday, August 02, 2006 4:02 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> >
> >
> > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
> > > Well yes, it looked dubious to me too, although I can't
> > find the syntaxt documented anywhere.
> > > However, that's what DUNDis giving me as a path to the phone!
> > >
> > > Something is screwed with DUNDi and SIP. Has ANYONE
> > actually implemnted it?
> > > I can't find it documented anywhere
> > >
> > > Doug.
> >
> > DUNDi gives you only what you give it to give you. You're the one
> > that needs to set the dial string correctly in DUNDi to get
> one back
> > that works. DUNDi is only as automatic as you let it be.
> >
> > This is what ours looks like. We don't use the iax
> versions (mainly
> > cause I want a homogenous SIP system), but we have entries
> in sip.conf
>
> > include files for each of the servers so we just dial
> > ${server}/${number}. This has been working for us for
> about 2 months
> > now, pretty much flawlessly as long as the phone's registered.
> >
> > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax =>
> > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax =>
> > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> >
> > [scm1]
> > type=friend
> > secret=p4ssw0rd
> > insecure=very
> > context=incoming
> > host=scm1.shsu.edu
> > qualify=yes
> > nat=no
>
> Aaron, while not really sure what I was doing, but extending
> beyond your
> example, I gave this a shot:
>
> dundi.conf:
> 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
> 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
>
> sip.conf:
> [dundisip1]
> type=friend
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx1.ipt.twoeighty.com
> qualify=yes
>
> [dundisip2]
> type=friend
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx2.ipt.twoeighty.com
> qualify=yes
>
> A CLI lookup looks better...
>
> *CLI> dundi lookup 9220370@180netsip
> 1. 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
> from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup completed in
> 107 ms
>
> However, my dial still fails...
>
> [Aug 2 16:42:44] -- AGI Script Executing Application: (Dial)
> Options: (SIP/dundisip2/9220370)
> [Aug 2 16:42:44] -- Called dundisip2/9220370
> [Aug 2 16:42:44] NOTICE[10474]: chan_sip.c:9685
> handle_response_invite:
> Failed to authenticate on INVITE to '"Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1'
> [Aug 2 16:42:44] -- SIP/dundisip2-7b30 is circuit-busy
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> The contents of this e-mail are intended for the named
> addressee only. It contains information that may be
> confidential. Unless you are the named addressee or an
> authorized designee, you may not copy or use it, or disclose
> it to anyone else. If you received it in error please notify
> us immediately and then destroy it.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
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asterisk-users mailing list
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The contents of this e-mail are intended for the named addressee only. It
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to anyone else. If you received it in error please notify us immediately and
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Thanks Bradley.
Here's a full sip console trace on the first pbx box.
xxx.yyy.128.18(phone 3254101): Originating phone, registered on pbx1.
xxx.yyy.142.162: pbx1
xxx.yyy.142.163: pbx2
Do you know what you are looking for?
Doug.
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>
CSeq: 1 INVITE
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 229
v=0
o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18
s=Polycom IP Phone
c=IN IP4 xxx.yyy.128.18
t=0 0
a=sendrecv
m=audio 2260 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
[Aug 3 09:40:28] --- (14 headers 10 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Using INVITE request as basis request -
86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
[Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT)
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Proxy-Authenticate: Digest realm="ipt.twoeighty.com",
nonce="673c57e6"
Content-Length: 0
---
[Aug 3 09:40:28] Scheduling destruction of call
'86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18' in 15000 ms
[Aug 3 09:40:28] Found user '3254101'
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
ACK sip:9220370@labpbx1.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f
CSeq: 1 ACK
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0
[Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>
CSeq: 2 INVITE
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="3254101",
realm="ipt.twoeighty.com", nonce="673c57e6",
uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone",
response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 229
v=0
o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18
s=Polycom IP Phone
c=IN IP4 xxx.yyy.128.18
t=0 0
a=sendrecv
m=audio 2260 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
[Aug 3 09:40:28] --- (15 headers 10 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Using INVITE request as basis request -
86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
[Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT)
[Aug 3 09:40:28] Found user '3254101'
[Aug 3 09:40:28] Found RTP audio format 0
[Aug 3 09:40:28] Found RTP audio format 18
[Aug 3 09:40:28] Found RTP audio format 101
[Aug 3 09:40:28] Peer audio RTP is at port xxx.yyy.128.18:2260
[Aug 3 09:40:28] Found description format PCMU
[Aug 3 09:40:28] Found description format G729
[Aug 3 09:40:28] Found description format telephone-event
[Aug 3 09:40:28] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104
(ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
[Aug 3 09:40:28] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
[Aug 3 09:40:28] Looking for 9220370 in pbx_betty_start (domain
labpbx1.ipt.twoeighty.com)
[Aug 3 09:40:28] list_route: hop: <sip:3254101@xxx.yyy.128.18>
[Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Content-Length: 0
---
[Aug 3 09:40:28] -- Executing NoOp("SIP/3254101-3ebc", "***
OnNet originated call "Chocolate Chip" <3254101> ->
9220370") in new stack
[Aug 3 09:40:28] -- Executing AGI("SIP/3254101-3ebc",
"ipt/originator.py") in new stack
[Aug 3 09:40:28] -- Launched AGI Script
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug 3 09:40:28] -- AGI Script Executing Application: (SetAccount) Options:
(9220370)
[Aug 3 09:40:28] -- AGI Script Executing Application: (ChanIsAvail)
Options: (SIP/9220370)
[Aug 3 09:40:28] Destroying call
'46b578613a629ebb69171e8b7f9b8458@xxx.yyy.142.162'
[Aug 3 09:40:28] -- AGI Script Executing Application: (Dial) Options:
(SIP/dundisip2/9220370)
[Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164
[Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP
[Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP
[Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 3 09:40:28] 14 headers, 12 lines
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.142.163:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13082 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Aug 3 09:40:28] -- Called dundisip2/9220370
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.142.163:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Proxy-Authenticate: Digest realm="ipt.twoeighty.com",
nonce="122095ae"
Content-Length: 0
[Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
---
[Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164
[Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP
[Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP
[Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.142.163:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Proxy-Authorization: Digest username="dundisip2",
realm="ipt.twoeighty.com", algorithm=MD5,
uri="sip:9220370@labpbx2.ipt.twoeighty.com",
nonce="122095ae",
response="c3812ae6a639df3827b26ff969acad23", opaque=""
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13083 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.142.163:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Content-Length: 0
[Aug 3 09:40:28] --- (10 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
---
[Aug 3 09:40:28] WARNING[13063]: chan_sip.c:9696 handle_response_invite:
Forbidden - wrong password on authentication for INVITE to '"Chocolate
Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86'
[Aug 3 09:40:28] -- SIP/dundisip2-936f is circuit-busy
[Aug 3 09:40:28] == Everyone is busy/congested at this time (1:0/1/0)
[Aug 3 09:40:28] -- AGI Script Executing Application: (Dial) Options:
(Local/u9220370@global_vmdeposit)
[Aug 3 09:40:28] -- Called u9220370@global_vmdeposit
[Aug 3 09:40:28] -- Executing
Answer("Local/u9220370@global_vmdeposit-3666,2", "") in new
stack
[Aug 3 09:40:28] -- Local/u9220370@global_vmdeposit-3666,1 answered
SIP/3254101-3ebc
[Aug 3 09:40:28] -- Executing
Wait("Local/u9220370@global_vmdeposit-3666,2", "1") in new
stack
[Aug 3 09:40:28] We're at xxx.yyy.142.162 port 28718
[Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP
[Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP
[Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13082 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 28718 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
ACK sip:9220370@xxx.yyy.142.162 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKfc84778d745812F4
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
CSeq: 2 ACK
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0
[Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Destroying call
'7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162'
[Aug 3 09:40:29] == Spawn extension (pbx_betty_start, 9220370, 2) exited
non-zero on 'Local/u9220370@global_vmdeposit-3666,2<ZOMBIE>'
[Aug 3 09:40:29] -- Executing VoiceMail("SIP/3254101-3ebc",
"u9220370@voicemail") in new stack
uniqueid => 58
customer_id => 0
context => voicemail
mailbox => 9220370
password => 1234
operator => No
attach => Yes
delete => No
stamp => 0000-00-00 00:00:00
envelope => Yes
saycid => Yes
[Aug 3 09:40:29] -- Playing 'vm-theperson' (language 'en')
[Aug 3 09:40:30]
<-- SIP read from xxx.yyy.128.18:5060:
BYE sip:9220370@xxx.yyy.142.162 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
CSeq: 3 BYE
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Proxy-Authorization: Digest username="3254101",
realm="ipt.twoeighty.com", nonce="673c57e6",
uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone",
response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
[Aug 3 09:40:30] --- (11 headers 0 lines)[Aug 3 09:40:30] ---
[Aug 3 09:40:30] Sending to xxx.yyy.128.18 : 5060 (non-NAT)
[Aug 3 09:40:30] Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Content-Length: 0
---
[Aug 3 09:40:30] == Spawn extension (global_vmdeposit, u9220370, 3) exited
non-zero on 'SIP/3254101-3ebc'
[Aug 3 09:40:31] Destroying call
'86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18'
and here's a full sip console trace on the second pbx box.
<-- SIP read from xxx.yyy.142.162:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13082 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 12 lines)---
Using INVITE request as basis request -
7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
Sending to xxx.yyy.142.162 : 5060 (NAT)
Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Proxy-Authenticate: Digest realm="ipt.twoeighty.com",
nonce="122095ae"
Content-Length: 0
---
Scheduling destruction of call
'7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' in 15000 ms
Found user '3254101'
<-- SIP read from xxx.yyy.142.162:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
--- (11 headers 0 lines)---
<-- SIP read from xxx.yyy.142.162:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Proxy-Authorization: Digest username="dundisip2",
realm="ipt.twoeighty.com", algorithm=MD5,
uri="sip:9220370@labpbx2.ipt.twoeighty.com",
nonce="122095ae",
response="c3812ae6a639df3827b26ff969acad23", opaque=""
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13083 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 12 lines)---
Using INVITE request as basis request -
7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
Sending to xxx.yyy.142.162 : 5060 (NAT)
Found user '3254101'
Aug 3 09:40:25 NOTICE[10884]: chan_sip.c:10469 handle_request_invite: Failed to
authenticate user "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Content-Length: 0
---
<-- SIP read from xxx.yyy.142.162:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162'
> -----Original Message-----
> From: Watkins, Bradley
> [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Watkins,
> Bradley
> Sent: Wednesday, August 02, 2006 6:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> Could you perhaps post a sip debug from both of the Asterisk
> consoles for the respective peers?
> ?
> I think we're very close, and I definitely want to get this
> working for you.
> ?
> Regards,
> - Brad
>
> _____
>
> From: asterisk-users-bounces@lists.digium.com on behalf of
> Douglas Garstang
> Sent: Wed 8/2/2006 7:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
>
> Bradley,
>
> I changed the type from friend to peer in sip.conf...
>
> [dundisip1]
> type=peer
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx1.ipt.twoeighty.com
> qualify=yes
>
> [dundisip2]
> type=peer
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx2.ipt.twoeighty.com
> qualify=yes
>
> but that just yielded the same error...
>
> [Aug? 2 17:07:51] NOTICE[10971]: chan_sip.c:9685
> handle_response_invite: Failed to authenticate on INVITE to
> '"Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as5e6e3efe'
>
> Btw, here's a sip trace between the asterisk boxes...
>
> Capturing on eth0
> 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status:
> 407 Proxy Authentication Required
> 3?? 0.177497 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@labpbx1.ipt.oneeighty.com
> 4?? 0.329036 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 5?? 0.329492 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
> 6?? 0.643513 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP
> Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with
> session description
> 7?? 0.644104 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status:
> 407 Proxy Authentication Required
> 8?? 0.644356 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request:
> ACK sip:9220370@labpbx2.ipt.oneeighty.com
> 9?? 0.647990 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP
> Status: 200 OK, with session description
> 10?? 0.806136 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@xxx.yyy.142.162
> 11?? 2.280664 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> BYE sip:9220370@xxx.yyy.142.162
> 12?? 2.280780 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 200 OK
>
> You can see that the first system doesn't resend the INVITE
> with the auth credentials as requested.
>
> So, then I also put the username field, and sip.conf now
> looks like this:
>
> [dundisip1]
> type=peer
> username=dundisip1
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx1.ipt.twoeighty.com
> qualify=yes
>
> [dundisip2]
> type=peer
> username=dundisip2
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx2.ipt.twoeeighty.com
> qualify=yes
>
> and with dundi.conf as:
>
> 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
> 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
>
> A CLI lookup still yields:
>
> *CLI> dundi lookup 9220370@180netsip
> ? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
> ???? from 00:14:22:1e:2a:d0, expires in 0 s
> DUNDi lookup completed in 130 ms
>
> HOWEVER, at attempt to dial results in this now:
>
> [Aug? 2 17:14:00] WARNING[11178]: chan_sip.c:9696
> handle_response_invite: Forbidden - wrong password on
> authentication for INVITE to '"Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f'
>
> And here's the SIP trace for THAT!
>
> 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status:
> 407 Proxy Authentication Required
> 3?? 0.158008 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@labpbx1.ipt.twoeighty.com
> 4?? 0.295054 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 5?? 0.300557 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
> 6?? 0.602324 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP
> Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with
> session description
> 7?? 0.603002 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status:
> 407 Proxy Authentication Required
> 8?? 0.603303 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request:
> ACK sip:9220370@labpbx2.ipt.twoeighty.com
> 9?? 0.603485 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP
> Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with
> session description
> 10?? 0.604251 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status:
> 403 Forbidden
> 11?? 0.604553 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request:
> ACK sip:9220370@labpbx2.ipt.twoeighty.com
> 12?? 0.608324 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP
> Status: 200 OK, with session description
> 13?? 0.749668 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@xxx.yyy.142.162
>
> And on the second Asterisk console we have logged:
>
> Aug? 2 17:13:57 NOTICE[29764]: chan_sip.c:10469
> handle_request_invite: Failed to authenticate user "Chocolate
> Chip" <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f
>
> Ugh... not having much luck with this...
>
>
>
>
>
>
>
>
>
>
>
> > -----Original Message-----
> > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
> > Sent: Wednesday, August 02, 2006 5:01 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> >
> >
> > Try putting a username= in the peer (BTW, use peer not friend)
> > definitions.? You appear to be attempting to authenticate as the
> > originating callerid (3254101).
> >
> > - Brad
> >
> > -----Original Message-----
> > From: asterisk-users-bounces@lists.digium.com
> > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf
> Of Douglas
> > Garstang
> > Sent: Wednesday, August 02, 2006 6:45 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> >
> > > -----Original Message-----
> > > From: Aaron Daniel [mailto:amdtech@shsu.edu]
> > > Sent: Wednesday, August 02, 2006 4:02 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> > >
> > >
> > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
> > > > Well yes, it looked dubious to me too, although I can't
> > > find the syntaxt documented anywhere.
> > > > However, that's what DUNDis giving me as a path to the
phone!
> > > >
> > > > Something is screwed with DUNDi and SIP. Has ANYONE
> > > actually implemnted it?
> > > > I can't find it documented anywhere
> > > >
> > > > Doug.
> > >
> > > DUNDi gives you only what you give it to give you.? You're
the one
> > > that needs to set the dial string correctly in DUNDi to get
> > one back
> > > that works.? DUNDi is only as automatic as you let it be.
> > >
> > > This is what ours looks like.? We don't use the iax
> > versions (mainly
> > > cause I want a homogenous SIP system), but we have entries
> > in sip.conf
> >
> > > include files for each of the servers so we just dial
> > > ${server}/${number}.? This has been working for us for
> > about 2 months
> > > now, pretty much flawlessly as long as the phone's
registered.
> > >
> > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax =>
> > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> > > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax
=>
> > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> > >
> > > [scm1]
> > > type=friend
> > > secret=p4ssw0rd
> > > insecure=very
> > > context=incoming
> > > host=scm1.shsu.edu
> > > qualify=yes
> > > nat=no
> >
> > Aaron, while not really sure what I was doing, but extending
> > beyond your
> > example, I gave this a shot:
> >
> > dundi.conf:
> > 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
> > 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
> >
> > sip.conf:
> > [dundisip1]
> > type=friend
> > secret=password
> > insecure=very
> > context=global_dundi_local
> > host=labpbx1.ipt.twoeighty.com
> > qualify=yes
> >
> > [dundisip2]
> > type=friend
> > secret=password
> > insecure=very
> > context=global_dundi_local
> > host=labpbx2.ipt.twoeighty.com
> > qualify=yes
> >
> > A CLI lookup looks better...
> >
> > *CLI> dundi lookup 9220370@180netsip
> >?? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
> >????? from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup
> completed in
> > 107 ms
> >
> > However, my dial still fails...
> >
> > [Aug? 2 16:42:44]???? -- AGI Script Executing Application: (Dial)
> > Options: (SIP/dundisip2/9220370)
> > [Aug? 2 16:42:44]???? -- Called dundisip2/9220370
> > [Aug? 2 16:42:44] NOTICE[10474]: chan_sip.c:9685
> > handle_response_invite:
> > Failed to authenticate on INVITE to '"Chocolate Chip"
> > <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1'
> > [Aug? 2 16:42:44]???? -- SIP/dundisip2-7b30 is circuit-busy
> >
> >
> > _______________________________________________
> > --Bandwidth and Colocation provided by Easynews.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >??? http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > The contents of this e-mail are intended for the named
> > addressee only. It contains information that may be
> > confidential. Unless you are the named addressee or an
> > authorized designee, you may not copy or use it, or disclose
> > it to anyone else. If you received it in error please notify
> > us immediately and then destroy it.
> > _______________________________________________
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> >
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> >
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> ?? http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> The contents of this e-mail are intended for the named
> addressee only. It contains information that may be
> confidential. Unless you are the named addressee or an
> authorized designee, you may not copy or use it, or disclose
> it to anyone else. If you received it in error please notify
> us immediately and then destroy it.
>
I'm basically trying to figure out why you aren't authenticating
properly.
When you posted the [peer] sections of your sip.conf, were the secret= lines
literal, or did you replace the actual secret with "password"?
- Brad
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang
Sent: Thursday, August 03, 2006 11:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Thanks Bradley.
Here's a full sip console trace on the first pbx box.
xxx.yyy.128.18(phone 3254101): Originating phone, registered on pbx1.
xxx.yyy.142.162: pbx1
xxx.yyy.142.163: pbx2
Do you know what you are looking for?
Doug.
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>
CSeq: 1 INVITE
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 229
v=0
o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18 s=Polycom IP Phone c=IN IP4
xxx.yyy.128.18 t=0 0 a=sendrecv m=audio 2260 RTP/AVP 0 18 101 a=rtpmap:0
PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
[Aug 3 09:40:28] --- (14 headers 10 lines)[Aug 3 09:40:28] --- [Aug 3
09:40:28] Using INVITE request as basis request -
86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
[Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT) [Aug 3 09:40:28]
Reliably Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Proxy-Authenticate: Digest realm="ipt.twoeighty.com",
nonce="673c57e6"
Content-Length: 0
---
[Aug 3 09:40:28] Scheduling destruction of call
'86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18' in 15000 ms [Aug 3
09:40:28] Found user '3254101'
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
ACK sip:9220370@labpbx1.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f
CSeq: 1 ACK
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0
[Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>
CSeq: 2 INVITE
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="3254101",
realm="ipt.twoeighty.com", nonce="673c57e6",
uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone",
response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 229
v=0
o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18
s=Polycom IP Phone
c=IN IP4 xxx.yyy.128.18
t=0 0
a=sendrecv
m=audio 2260 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
[Aug 3 09:40:28] --- (15 headers 10 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Using INVITE request as basis request -
86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
[Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT)
[Aug 3 09:40:28] Found user '3254101'
[Aug 3 09:40:28] Found RTP audio format 0
[Aug 3 09:40:28] Found RTP audio format 18
[Aug 3 09:40:28] Found RTP audio format 101
[Aug 3 09:40:28] Peer audio RTP is at port xxx.yyy.128.18:2260
[Aug 3 09:40:28] Found description format PCMU
[Aug 3 09:40:28] Found description format G729
[Aug 3 09:40:28] Found description format telephone-event
[Aug 3 09:40:28] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104
(ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
[Aug 3 09:40:28] Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
[Aug 3 09:40:28] Looking for 9220370 in pbx_betty_start (domain
labpbx1.ipt.twoeighty.com)
[Aug 3 09:40:28] list_route: hop: <sip:3254101@xxx.yyy.128.18>
[Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Content-Length: 0
---
[Aug 3 09:40:28] -- Executing NoOp("SIP/3254101-3ebc", "***
OnNet originated call "Chocolate Chip" <3254101> ->
9220370") in new stack
[Aug 3 09:40:28] -- Executing AGI("SIP/3254101-3ebc",
"ipt/originator.py") in new stack
[Aug 3 09:40:28] -- Launched AGI Script
/var/lib/asterisk/agi-bin/ipt/originator.py
[Aug 3 09:40:28] -- AGI Script Executing Application: (SetAccount) Options:
(9220370)
[Aug 3 09:40:28] -- AGI Script Executing Application: (ChanIsAvail)
Options: (SIP/9220370)
[Aug 3 09:40:28] Destroying call
'46b578613a629ebb69171e8b7f9b8458@xxx.yyy.142.162'
[Aug 3 09:40:28] -- AGI Script Executing Application: (Dial) Options:
(SIP/dundisip2/9220370)
[Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164
[Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP
[Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP
[Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 3 09:40:28] 14 headers, 12 lines
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.142.163:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13082 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Aug 3 09:40:28] -- Called dundisip2/9220370
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.142.163:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Proxy-Authenticate: Digest realm="ipt.twoeighty.com",
nonce="122095ae"
Content-Length: 0
[Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
---
[Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164
[Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP
[Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP
[Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.142.163:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Proxy-Authorization: Digest username="dundisip2",
realm="ipt.twoeighty.com", algorithm=MD5,
uri="sip:9220370@labpbx2.ipt.twoeighty.com",
nonce="122095ae",
response="c3812ae6a639df3827b26ff969acad23", opaque=""
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13083 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.142.163:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Content-Length: 0
[Aug 3 09:40:28] --- (10 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
---
[Aug 3 09:40:28] WARNING[13063]: chan_sip.c:9696 handle_response_invite:
Forbidden - wrong password on authentication for INVITE to '"Chocolate
Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86'
[Aug 3 09:40:28] -- SIP/dundisip2-936f is circuit-busy
[Aug 3 09:40:28] == Everyone is busy/congested at this time (1:0/1/0)
[Aug 3 09:40:28] -- AGI Script Executing Application: (Dial) Options:
(Local/u9220370@global_vmdeposit)
[Aug 3 09:40:28] -- Called u9220370@global_vmdeposit
[Aug 3 09:40:28] -- Executing
Answer("Local/u9220370@global_vmdeposit-3666,2", "") in new
stack
[Aug 3 09:40:28] -- Local/u9220370@global_vmdeposit-3666,1 answered
SIP/3254101-3ebc
[Aug 3 09:40:28] -- Executing
Wait("Local/u9220370@global_vmdeposit-3666,2", "1") in new
stack
[Aug 3 09:40:28] We're at xxx.yyy.142.162 port 28718
[Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP
[Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP
[Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 3 09:40:28] Reliably Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13082 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 28718 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
[Aug 3 09:40:28]
<-- SIP read from xxx.yyy.128.18:5060:
ACK sip:9220370@xxx.yyy.142.162 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKfc84778d745812F4
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
CSeq: 2 ACK
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0
[Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] ---
[Aug 3 09:40:28] Destroying call
'7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162'
[Aug 3 09:40:29] == Spawn extension (pbx_betty_start, 9220370, 2) exited
non-zero on 'Local/u9220370@global_vmdeposit-3666,2<ZOMBIE>'
[Aug 3 09:40:29] -- Executing VoiceMail("SIP/3254101-3ebc",
"u9220370@voicemail") in new stack
uniqueid => 58
customer_id => 0
context => voicemail
mailbox => 9220370
password => 1234
operator => No
attach => Yes
delete => No
stamp => 0000-00-00 00:00:00
envelope => Yes
saycid => Yes
[Aug 3 09:40:29] -- Playing 'vm-theperson' (language 'en')
[Aug 3 09:40:30]
<-- SIP read from xxx.yyy.128.18:5060:
BYE sip:9220370@xxx.yyy.142.162 SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
CSeq: 3 BYE
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
Contact: <sip:3254101@xxx.yyy.128.18>
User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067
Proxy-Authorization: Digest username="3254101",
realm="ipt.twoeighty.com", nonce="673c57e6",
uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone",
response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5
Max-Forwards: 70
Content-Length: 0
[Aug 3 09:40:30] --- (11 headers 0 lines)[Aug 3 09:40:30] ---
[Aug 3 09:40:30] Sending to xxx.yyy.128.18 : 5060 (non-NAT)
[Aug 3 09:40:30] Transmitting (no NAT) to xxx.yyy.128.18:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996;received=xxx.yyy.128.18
From: "Chocolate Chip"
<sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC
To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6
Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.162>
Content-Length: 0
---
[Aug 3 09:40:30] == Spawn extension (global_vmdeposit, u9220370, 3) exited
non-zero on 'SIP/3254101-3ebc'
[Aug 3 09:40:31] Destroying call
'86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18'
and here's a full sip console trace on the second pbx box.
<-- SIP read from xxx.yyy.142.162:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13082 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (14 headers 12 lines)---
Using INVITE request as basis request -
7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
Sending to xxx.yyy.142.162 : 5060 (NAT)
Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Proxy-Authenticate: Digest realm="ipt.twoeighty.com",
nonce="122095ae"
Content-Length: 0
---
Scheduling destruction of call
'7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' in 15000 ms
Found user '3254101'
<-- SIP read from xxx.yyy.142.162:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
--- (11 headers 0 lines)---
<-- SIP read from xxx.yyy.142.162:5060:
INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Proxy-Authorization: Digest username="dundisip2",
realm="ipt.twoeighty.com", algorithm=MD5,
uri="sip:9220370@labpbx2.ipt.twoeighty.com",
nonce="122095ae",
response="c3812ae6a639df3827b26ff969acad23", opaque=""
Date: Thu, 03 Aug 2006 15:40:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269
v=0
o=root 13082 13083 IN IP4 xxx.yyy.142.162
s=session
c=IN IP4 xxx.yyy.142.162
t=0 0
m=audio 27164 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
--- (15 headers 12 lines)---
Using INVITE request as basis request -
7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
Sending to xxx.yyy.142.162 : 5060 (NAT)
Found user '3254101'
Aug 3 09:40:25 NOTICE[10884]: chan_sip.c:10469 handle_request_invite: Failed to
authenticate user "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx.yyy.142.162
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:9220370@xxx.yyy.142.163>
Content-Length: 0
---
<-- SIP read from xxx.yyy.142.162:5060:
ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport
From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as097cbc86
To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd
Contact: <sip:3254101@xxx.yyy.142.162>
Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162'
> -----Original Message-----
> From: Watkins, Bradley
> [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Watkins,
> Bradley
> Sent: Wednesday, August 02, 2006 6:31 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
> Could you perhaps post a sip debug from both of the Asterisk
> consoles for the respective peers?
> ?
> I think we're very close, and I definitely want to get this
> working for you.
> ?
> Regards,
> - Brad
>
> _____
>
> From: asterisk-users-bounces@lists.digium.com on behalf of
> Douglas Garstang
> Sent: Wed 8/2/2006 7:16 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [asterisk-users] Re: DUNDi with SIP
>
>
>
> Bradley,
>
> I changed the type from friend to peer in sip.conf...
>
> [dundisip1]
> type=peer
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx1.ipt.twoeighty.com
> qualify=yes
>
> [dundisip2]
> type=peer
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx2.ipt.twoeighty.com
> qualify=yes
>
> but that just yielded the same error...
>
> [Aug? 2 17:07:51] NOTICE[10971]: chan_sip.c:9685
> handle_response_invite: Failed to authenticate on INVITE to
> '"Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as5e6e3efe'
>
> Btw, here's a sip trace between the asterisk boxes...
>
> Capturing on eth0
> 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status:
> 407 Proxy Authentication Required
> 3?? 0.177497 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@labpbx1.ipt.oneeighty.com
> 4?? 0.329036 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 5?? 0.329492 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
> 6?? 0.643513 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP
> Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with
> session description
> 7?? 0.644104 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status:
> 407 Proxy Authentication Required
> 8?? 0.644356 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request:
> ACK sip:9220370@labpbx2.ipt.oneeighty.com
> 9?? 0.647990 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP
> Status: 200 OK, with session description
> 10?? 0.806136 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@xxx.yyy.142.162
> 11?? 2.280664 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> BYE sip:9220370@xxx.yyy.142.162
> 12?? 2.280780 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 200 OK
>
> You can see that the first system doesn't resend the INVITE
> with the auth credentials as requested.
>
> So, then I also put the username field, and sip.conf now
> looks like this:
>
> [dundisip1]
> type=peer
> username=dundisip1
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx1.ipt.twoeighty.com
> qualify=yes
>
> [dundisip2]
> type=peer
> username=dundisip2
> secret=password
> insecure=very
> context=global_dundi_local
> host=labpbx2.ipt.twoeeighty.com
> qualify=yes
>
> and with dundi.conf as:
>
> 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
> 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
>
> A CLI lookup still yields:
>
> *CLI> dundi lookup 9220370@180netsip
> ? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
> ???? from 00:14:22:1e:2a:d0, expires in 0 s
> DUNDi lookup completed in 130 ms
>
> HOWEVER, at attempt to dial results in this now:
>
> [Aug? 2 17:14:00] WARNING[11178]: chan_sip.c:9696
> handle_response_invite: Forbidden - wrong password on
> authentication for INVITE to '"Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f'
>
> And here's the SIP trace for THAT!
>
> 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status:
> 407 Proxy Authentication Required
> 3?? 0.158008 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@labpbx1.ipt.twoeighty.com
> 4?? 0.295054 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP
> Request: INVITE
> sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with
> session description
> 5?? 0.300557 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 100 Trying
> 6?? 0.602324 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP
> Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with
> session description
> 7?? 0.603002 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status:
> 407 Proxy Authentication Required
> 8?? 0.603303 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request:
> ACK sip:9220370@labpbx2.ipt.twoeighty.com
> 9?? 0.603485 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP
> Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with
> session description
> 10?? 0.604251 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status:
> 403 Forbidden
> 11?? 0.604553 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request:
> ACK sip:9220370@labpbx2.ipt.twoeighty.com
> 12?? 0.608324 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP
> Status: 200 OK, with session description
> 13?? 0.749668 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request:
> ACK sip:9220370@xxx.yyy.142.162
>
> And on the second Asterisk console we have logged:
>
> Aug? 2 17:13:57 NOTICE[29764]: chan_sip.c:10469
> handle_request_invite: Failed to authenticate user "Chocolate
> Chip" <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f
>
> Ugh... not having much luck with this...
>
>
>
>
>
>
>
>
>
>
>
> > -----Original Message-----
> > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
> > Sent: Wednesday, August 02, 2006 5:01 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> >
> >
> > Try putting a username= in the peer (BTW, use peer not friend)
> > definitions.? You appear to be attempting to authenticate as the
> > originating callerid (3254101).
> >
> > - Brad
> >
> > -----Original Message-----
> > From: asterisk-users-bounces@lists.digium.com
> > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf
> Of Douglas
> > Garstang
> > Sent: Wednesday, August 02, 2006 6:45 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> >
> > > -----Original Message-----
> > > From: Aaron Daniel [mailto:amdtech@shsu.edu]
> > > Sent: Wednesday, August 02, 2006 4:02 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: RE: [asterisk-users] Re: DUNDi with SIP
> > >
> > >
> > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote:
> > > > Well yes, it looked dubious to me too, although I can't
> > > find the syntaxt documented anywhere.
> > > > However, that's what DUNDis giving me as a path to the
phone!
> > > >
> > > > Something is screwed with DUNDi and SIP. Has ANYONE
> > > actually implemnted it?
> > > > I can't find it documented anywhere
> > > >
> > > > Doug.
> > >
> > > DUNDi gives you only what you give it to give you.? You're
the one
> > > that needs to set the dial string correctly in DUNDi to get
> > one back
> > > that works.? DUNDi is only as automatic as you let it be.
> > >
> > > This is what ours looks like.? We don't use the iax
> > versions (mainly
> > > cause I want a homogenous SIP system), but we have entries
> > in sip.conf
> >
> > > include files for each of the servers so we just dial
> > > ${server}/${number}.? This has been working for us for
> > about 2 months
> > > now, pretty much flawlessly as long as the phone's
registered.
> > >
> > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax =>
> > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> > > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax
=>
> > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER}
> > >
> > > [scm1]
> > > type=friend
> > > secret=p4ssw0rd
> > > insecure=very
> > > context=incoming
> > > host=scm1.shsu.edu
> > > qualify=yes
> > > nat=no
> >
> > Aaron, while not really sure what I was doing, but extending
> > beyond your
> > example, I gave this a shot:
> >
> > dundi.conf:
> > 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER}
> > 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER}
> >
> > sip.conf:
> > [dundisip1]
> > type=friend
> > secret=password
> > insecure=very
> > context=global_dundi_local
> > host=labpbx1.ipt.twoeighty.com
> > qualify=yes
> >
> > [dundisip2]
> > type=friend
> > secret=password
> > insecure=very
> > context=global_dundi_local
> > host=labpbx2.ipt.twoeighty.com
> > qualify=yes
> >
> > A CLI lookup looks better...
> >
> > *CLI> dundi lookup 9220370@180netsip
> >?? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH)
> >????? from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup
> completed in
> > 107 ms
> >
> > However, my dial still fails...
> >
> > [Aug? 2 16:42:44]???? -- AGI Script Executing Application: (Dial)
> > Options: (SIP/dundisip2/9220370)
> > [Aug? 2 16:42:44]???? -- Called dundisip2/9220370
> > [Aug? 2 16:42:44] NOTICE[10474]: chan_sip.c:9685
> > handle_response_invite:
> > Failed to authenticate on INVITE to '"Chocolate Chip"
> > <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1'
> > [Aug? 2 16:42:44]???? -- SIP/dundisip2-7b30 is circuit-busy
> >
> >
> > _______________________________________________
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Brad, Not sure what you mean... I had literally 'secret=password'.... password being 'password' for testing purposes...> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Thursday, August 03, 2006 12:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > I'm basically trying to figure out why you aren't > authenticating properly. > > When you posted the [peer] sections of your sip.conf, were > the secret= lines literal, or did you replace the actual > secret with "password"? > > - Brad > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Douglas Garstang > Sent: Thursday, August 03, 2006 11:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > Thanks Bradley. > > Here's a full sip console trace on the first pbx box. > > xxx.yyy.128.18(phone 3254101): Originating phone, registered on pbx1. > xxx.yyy.142.162: pbx1 > xxx.yyy.142.163: pbx2 > > Do you know what you are looking for? > > Doug. > > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone> > CSeq: 1 INVITE > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Supported: 100rel,replace > Allow-Events: talk,hold,conference > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 229 > > v=0 > o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18 s=Polycom IP > Phone c=IN IP4 xxx.yyy.128.18 t=0 0 a=sendrecv m=audio 2260 > RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > > [Aug 3 09:40:28] --- (14 headers 10 lines)[Aug 3 09:40:28] > --- [Aug 3 09:40:28] Using INVITE request as basis request - > 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > [Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT) > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.128.18:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Proxy-Authenticate: Digest realm="ipt.twoeighty.com", nonce="673c57e6" > Content-Length: 0 > > > --- > [Aug 3 09:40:28] Scheduling destruction of call > '86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18' in 15000 ms [Aug > 3 09:40:28] Found user '3254101' > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > ACK sip:9220370@labpbx1.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f > CSeq: 1 ACK > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Max-Forwards: 70 > Content-Length: 0 > > > [Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone> > CSeq: 2 INVITE > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Supported: 100rel,replace > Allow-Events: talk,hold,conference > Proxy-Authorization: Digest username="3254101", > realm="ipt.twoeighty.com", nonce="673c57e6", > uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone", > response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5 > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 229 > > v=0 > o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18 > s=Polycom IP Phone > c=IN IP4 xxx.yyy.128.18 > t=0 0 > a=sendrecv > m=audio 2260 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > > [Aug 3 09:40:28] --- (15 headers 10 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Using INVITE request as basis request - > 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > [Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT) > [Aug 3 09:40:28] Found user '3254101' > [Aug 3 09:40:28] Found RTP audio format 0 > [Aug 3 09:40:28] Found RTP audio format 18 > [Aug 3 09:40:28] Found RTP audio format 101 > [Aug 3 09:40:28] Peer audio RTP is at port xxx.yyy.128.18:2260 > [Aug 3 09:40:28] Found description format PCMU > [Aug 3 09:40:28] Found description format G729 > [Aug 3 09:40:28] Found description format telephone-event > [Aug 3 09:40:28] Capabilities: us - 0x104 (ulaw|g729), peer > - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - > 0x104 (ulaw|g729) > [Aug 3 09:40:28] Non-codec capabilities: us - 0x1 > (telephone-event), peer - 0x1 (telephone-event), combined - > 0x1 (telephone-event) > [Aug 3 09:40:28] Looking for 9220370 in pbx_betty_start > (domain labpbx1.ipt.twoeighty.com) > [Aug 3 09:40:28] list_route: hop: <sip:3254101@xxx.yyy.128.18> > [Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.128.18:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone> > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Content-Length: 0 > > > --- > [Aug 3 09:40:28] -- Executing NoOp("SIP/3254101-3ebc", > "*** OnNet originated call "Chocolate Chip" <3254101> -> > 9220370") in new stack > [Aug 3 09:40:28] -- Executing AGI("SIP/3254101-3ebc", > "ipt/originator.py") in new stack > [Aug 3 09:40:28] -- Launched AGI Script > /var/lib/asterisk/agi-bin/ipt/originator.py > [Aug 3 09:40:28] -- AGI Script Executing Application: > (SetAccount) Options: (9220370) > [Aug 3 09:40:28] -- AGI Script Executing Application: > (ChanIsAvail) Options: (SIP/9220370) > [Aug 3 09:40:28] Destroying call > '46b578613a629ebb69171e8b7f9b8458@xxx.yyy.142.162' > [Aug 3 09:40:28] -- AGI Script Executing Application: > (Dial) Options: (SIP/dundisip2/9220370) > [Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164 > [Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP > [Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP > [Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP > [Aug 3 09:40:28] 14 headers, 12 lines > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.142.163:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13082 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > [Aug 3 09:40:28] -- Called dundisip2/9220370 > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.142.163:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Proxy-Authenticate: Digest realm="ipt.twoeighty.com", nonce="122095ae" > Content-Length: 0 > > > [Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- > [Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164 > [Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP > [Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP > [Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.142.163:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="122095ae", > response="c3812ae6a639df3827b26ff969acad23", opaque="" > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13083 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.142.163:5060: > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Content-Length: 0 > > > [Aug 3 09:40:28] --- (10 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- > [Aug 3 09:40:28] WARNING[13063]: chan_sip.c:9696 > handle_response_invite: Forbidden - wrong password on > authentication for INVITE to '"Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86' > [Aug 3 09:40:28] -- SIP/dundisip2-936f is circuit-busy > [Aug 3 09:40:28] == Everyone is busy/congested at this > time (1:0/1/0) > [Aug 3 09:40:28] -- AGI Script Executing Application: > (Dial) Options: (Local/u9220370@global_vmdeposit) > [Aug 3 09:40:28] -- Called u9220370@global_vmdeposit > [Aug 3 09:40:28] -- Executing > Answer("Local/u9220370@global_vmdeposit-3666,2", "") in new stack > [Aug 3 09:40:28] -- > Local/u9220370@global_vmdeposit-3666,1 answered SIP/3254101-3ebc > [Aug 3 09:40:28] -- Executing > Wait("Local/u9220370@global_vmdeposit-3666,2", "1") in new stack > [Aug 3 09:40:28] We're at xxx.yyy.142.162 port 28718 > [Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP > [Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP > [Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.128.18:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13082 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 28718 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > ACK sip:9220370@xxx.yyy.142.162 SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKfc84778d745812F4 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > CSeq: 2 ACK > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Max-Forwards: 70 > Content-Length: 0 > > > [Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Destroying call > '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' > [Aug 3 09:40:29] == Spawn extension (pbx_betty_start, > 9220370, 2) exited non-zero on > 'Local/u9220370@global_vmdeposit-3666,2<ZOMBIE>' > [Aug 3 09:40:29] -- Executing > VoiceMail("SIP/3254101-3ebc", "u9220370@voicemail") in new stack > uniqueid => 58 > customer_id => 0 > context => voicemail > mailbox => 9220370 > password => 1234 > operator => No > attach => Yes > delete => No > stamp => 0000-00-00 00:00:00 > envelope => Yes > saycid => Yes > [Aug 3 09:40:29] -- Playing 'vm-theperson' (language 'en') > [Aug 3 09:40:30] > <-- SIP read from xxx.yyy.128.18:5060: > BYE sip:9220370@xxx.yyy.142.162 SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > CSeq: 3 BYE > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Proxy-Authorization: Digest username="3254101", > realm="ipt.twoeighty.com", nonce="673c57e6", > uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone", > response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5 > Max-Forwards: 70 > Content-Length: 0 > > > [Aug 3 09:40:30] --- (11 headers 0 lines)[Aug 3 09:40:30] --- > [Aug 3 09:40:30] Sending to xxx.yyy.128.18 : 5060 (non-NAT) > [Aug 3 09:40:30] Transmitting (no NAT) to xxx.yyy.128.18:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 3 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Content-Length: 0 > > > --- > [Aug 3 09:40:30] == Spawn extension (global_vmdeposit, > u9220370, 3) exited non-zero on 'SIP/3254101-3ebc' > [Aug 3 09:40:31] Destroying call > '86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18' > > > and here's a full sip console trace on the second pbx box. > > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13082 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- (14 headers 12 lines)--- > Using INVITE request as basis request - > 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > Sending to xxx.yyy.142.162 : 5060 (NAT) > Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Proxy-Authenticate: Digest realm="ipt.twoeighty.com", nonce="122095ae" > Content-Length: 0 > > > --- > Scheduling destruction of call > '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' in 15000 ms > Found user '3254101' > > <-- SIP read from xxx.yyy.142.162:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- (11 headers 0 lines)--- > > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="122095ae", > response="c3812ae6a639df3827b26ff969acad23", opaque="" > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13083 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- (15 headers 12 lines)--- > Using INVITE request as basis request - > 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > Sending to xxx.yyy.142.162 : 5060 (NAT) > Found user '3254101' > Aug 3 09:40:25 NOTICE[10884]: chan_sip.c:10469 > handle_request_invite: Failed to authenticate user "Chocolate > Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060: > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Content-Length: 0 > > > --- > > <-- SIP read from xxx.yyy.142.162:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- (11 headers 0 lines)--- > Destroying call '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' > > > > > -----Original Message----- > > From: Watkins, Bradley > > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf > Of Watkins, > > Bradley > > Sent: Wednesday, August 02, 2006 6:31 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > Could you perhaps post a sip debug from both of the Asterisk > > consoles for the respective peers? > > ? > > I think we're very close, and I definitely want to get this > > working for you. > > ? > > Regards, > > - Brad > > > > _____ > > > > From: asterisk-users-bounces@lists.digium.com on behalf of > > Douglas Garstang > > Sent: Wed 8/2/2006 7:16 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > Bradley, > > > > I changed the type from friend to peer in sip.conf... > > > > [dundisip1] > > type=peer > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.twoeighty.com > > qualify=yes > > > > [dundisip2] > > type=peer > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.twoeighty.com > > qualify=yes > > > > but that just yielded the same error... > > > > [Aug? 2 17:07:51] NOTICE[10971]: chan_sip.c:9685 > > handle_response_invite: Failed to authenticate on INVITE to > > '"Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as5e6e3efe' > > > > Btw, here's a sip trace between the asterisk boxes... > > > > Capturing on eth0 > > 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > > 407 Proxy Authentication Required > > 3?? 0.177497 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@labpbx1.ipt.oneeighty.com > > 4?? 0.329036 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 5?? 0.329492 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > 100 Trying > > 6?? 0.643513 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP > > Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with > > session description > > 7?? 0.644104 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: > > 407 Proxy Authentication Required > > 8?? 0.644356 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: > > ACK sip:9220370@labpbx2.ipt.oneeighty.com > > 9?? 0.647990 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP > > Status: 200 OK, with session description > > 10?? 0.806136 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@xxx.yyy.142.162 > > 11?? 2.280664 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > BYE sip:9220370@xxx.yyy.142.162 > > 12?? 2.280780 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 200 OK > > > > You can see that the first system doesn't resend the INVITE > > with the auth credentials as requested. > > > > So, then I also put the username field, and sip.conf now > > looks like this: > > > > [dundisip1] > > type=peer > > username=dundisip1 > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.twoeighty.com > > qualify=yes > > > > [dundisip2] > > type=peer > > username=dundisip2 > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.twoeeighty.com > > qualify=yes > > > > and with dundi.conf as: > > > > 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER} > > 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER} > > > > A CLI lookup still yields: > > > > *CLI> dundi lookup 9220370@180netsip > > ? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH) > > ???? from 00:14:22:1e:2a:d0, expires in 0 s > > DUNDi lookup completed in 130 ms > > > > HOWEVER, at attempt to dial results in this now: > > > > [Aug? 2 17:14:00] WARNING[11178]: chan_sip.c:9696 > > handle_response_invite: Forbidden - wrong password on > > authentication for INVITE to '"Chocolate Chip" > > <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f' > > > > And here's the SIP trace for THAT! > > > > 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > > 407 Proxy Authentication Required > > 3?? 0.158008 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@labpbx1.ipt.twoeighty.com > > 4?? 0.295054 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 5?? 0.300557 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > 100 Trying > > 6?? 0.602324 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP > > Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with > > session description > > 7?? 0.603002 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: > > 407 Proxy Authentication Required > > 8?? 0.603303 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: > > ACK sip:9220370@labpbx2.ipt.twoeighty.com > > 9?? 0.603485 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP > > Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with > > session description > > 10?? 0.604251 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: > > 403 Forbidden > > 11?? 0.604553 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: > > ACK sip:9220370@labpbx2.ipt.twoeighty.com > > 12?? 0.608324 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP > > Status: 200 OK, with session description > > 13?? 0.749668 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@xxx.yyy.142.162 > > > > And on the second Asterisk console we have logged: > > > > Aug? 2 17:13:57 NOTICE[29764]: chan_sip.c:10469 > > handle_request_invite: Failed to authenticate user "Chocolate > > Chip" <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f > > > > Ugh... not having much luck with this... > > > > > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > > > Sent: Wednesday, August 02, 2006 5:01 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > Try putting a username= in the peer (BTW, use peer not friend) > > > definitions.? You appear to be attempting to authenticate as the > > > originating callerid (3254101). > > > > > > - Brad > > > > > > -----Original Message----- > > > From: asterisk-users-bounces@lists.digium.com > > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf > > Of Douglas > > > Garstang > > > Sent: Wednesday, August 02, 2006 6:45 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > -----Original Message----- > > > > From: Aaron Daniel [mailto:amdtech@shsu.edu] > > > > Sent: Wednesday, August 02, 2006 4:02 PM > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > > > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote: > > > > > Well yes, it looked dubious to me too, although I can't > > > > find the syntaxt documented anywhere. > > > > > However, that's what DUNDis giving me as a path to the phone! > > > > > > > > > > Something is screwed with DUNDi and SIP. Has ANYONE > > > > actually implemnted it? > > > > > I can't find it documented anywhere > > > > > > > > > > Doug. > > > > > > > > DUNDi gives you only what you give it to give you.? > You're the one > > > > that needs to set the dial string correctly in DUNDi to get > > > one back > > > > that works.? DUNDi is only as automatic as you let it be. > > > > > > > > This is what ours looks like.? We don't use the iax > > > versions (mainly > > > > cause I want a homogenous SIP system), but we have entries > > > in sip.conf > > > > > > > include files for each of the servers so we just dial > > > > ${server}/${number}.? This has been working for us for > > > about 2 months > > > > now, pretty much flawlessly as long as the phone's registered. > > > > > > > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax => > > > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > > > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax => > > > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > > > > > > > [scm1] > > > > type=friend > > > > secret=p4ssw0rd > > > > insecure=very > > > > context=incoming > > > > host=scm1.shsu.edu > > > > qualify=yes > > > > nat=no > > > > > > Aaron, while not really sure what I was doing, but extending > > > beyond your > > > example, I gave this a shot: > > > > > > dundi.conf: > > > 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER} > > > 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER} > > > > > > sip.conf: > > > [dundisip1] > > > type=friend > > > secret=password > > > insecure=very > > > context=global_dundi_local > > > host=labpbx1.ipt.twoeighty.com > > > qualify=yes > > > > > > [dundisip2] > > > type=friend > > > secret=password > > > insecure=very > > > context=global_dundi_local > > > host=labpbx2.ipt.twoeighty.com > > > qualify=yes > > > > > > A CLI lookup looks better... > > > > > > *CLI> dundi lookup 9220370@180netsip > > >?? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH) > > >????? from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup > > completed in > > > 107 ms > > > > > > However, my dial still fails... > > > > > > [Aug? 2 16:42:44]???? -- AGI Script Executing Application: (Dial) > > > Options: (SIP/dundisip2/9220370) > > > [Aug? 2 16:42:44]???? -- Called dundisip2/9220370 > > > [Aug? 2 16:42:44] NOTICE[10474]: chan_sip.c:9685 > > > handle_response_invite: > > > Failed to authenticate on INVITE to '"Chocolate Chip" > > > <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1' > > > [Aug? 2 16:42:44]???? -- SIP/dundisip2-7b30 is circuit-busy > > > > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >??? http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > The contents of this e-mail are intended for the named > > > addressee only. It contains information that may be > > > confidential. Unless you are the named addressee or an > > > authorized designee, you may not copy or use it, or disclose > > > it to anyone else. If you received it in error please notify > > > us immediately and then destroy it. > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >??? http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > ?? http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > The contents of this e-mail are intended for the named > > addressee only. It contains information that may be > > confidential. Unless you are the named addressee or an > > authorized designee, you may not copy or use it, or disclose > > it to anyone else. If you received it in error please notify > > us immediately and then destroy it. > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > The contents of this e-mail are intended for the named > addressee only. It contains information that may be > confidential. Unless you are the named addressee or an > authorized designee, you may not copy or use it, or disclose > it to anyone else. If you received it in error please notify > us immediately and then destroy it. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Brad, Here's the invite that the first Asterisk box sends to the second Asterisk box. I see no reference to dundisip1 or dundisip2 in there... I'm not even sure that SIP can support using different From: and authid's? [Aug 3 12:45:49] Reliably Transmitting (no NAT) to xxx.yyy.142.163:5060: INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK280b3a7d;rport From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as45fbdb6f To: <sip:9220370@labpbx2.ipt.twoeighty.com> Contact: <sip:3254101@xxx.yyy.142.162> Call-ID: 508a683641d3034d46403e1747755722@xxx.yyy.142.162 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Remote-Party-ID: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no Date: Thu, 03 Aug 2006 18:45:49 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 269 v=0 o=root 16089 16089 IN IP4 xxx.yyy.142.162 s=session c=IN IP4 xxx.yyy.142.162 t=0 0 m=audio 32738 RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - And once again, here's sip.conf: [dundisip1] type=peer username=dundisip1 secret=password insecure=very context=global_dundi_local host=labpbx1.ipt.oneeighty.com qualify=yes [dundisip2] type=peer username=dundisip2 secret=password insecure=very context=global_dundi_local host=labpbx2.ipt.oneeighty.com qualify=yes> -----Original Message----- > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > Sent: Thursday, August 03, 2006 12:29 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > I'm basically trying to figure out why you aren't > authenticating properly. > > When you posted the [peer] sections of your sip.conf, were > the secret= lines literal, or did you replace the actual > secret with "password"? > > - Brad > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Douglas Garstang > Sent: Thursday, August 03, 2006 11:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > Thanks Bradley. > > Here's a full sip console trace on the first pbx box. > > xxx.yyy.128.18(phone 3254101): Originating phone, registered on pbx1. > xxx.yyy.142.162: pbx1 > xxx.yyy.142.163: pbx2 > > Do you know what you are looking for? > > Doug. > > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone> > CSeq: 1 INVITE > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Supported: 100rel,replace > Allow-Events: talk,hold,conference > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 229 > > v=0 > o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18 s=Polycom IP > Phone c=IN IP4 xxx.yyy.128.18 t=0 0 a=sendrecv m=audio 2260 > RTP/AVP 0 18 101 a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > > [Aug 3 09:40:28] --- (14 headers 10 lines)[Aug 3 09:40:28] > --- [Aug 3 09:40:28] Using INVITE request as basis request - > 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > [Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT) > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.128.18:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 1 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Proxy-Authenticate: Digest realm="ipt.twoeighty.com", nonce="673c57e6" > Content-Length: 0 > > > --- > [Aug 3 09:40:28] Scheduling destruction of call > '86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18' in 15000 ms [Aug > 3 09:40:28] Found user '3254101' > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > ACK sip:9220370@labpbx1.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKeac2a6619BA84D58 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as7757655f > CSeq: 1 ACK > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Max-Forwards: 70 > Content-Length: 0 > > > [Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > INVITE sip:9220370@labpbx1.ipt.twoeighty.com;user=phone SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone> > CSeq: 2 INVITE > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Supported: 100rel,replace > Allow-Events: talk,hold,conference > Proxy-Authorization: Digest username="3254101", > realm="ipt.twoeighty.com", nonce="673c57e6", > uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone", > response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5 > Max-Forwards: 70 > Content-Type: application/sdp > Content-Length: 229 > > v=0 > o=- 1154619309 1154619309 IN IP4 xxx.yyy.128.18 > s=Polycom IP Phone > c=IN IP4 xxx.yyy.128.18 > t=0 0 > a=sendrecv > m=audio 2260 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=rtpmap:101 telephone-event/8000 > > [Aug 3 09:40:28] --- (15 headers 10 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Using INVITE request as basis request - > 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > [Aug 3 09:40:28] Sending to xxx.yyy.128.18 : 5060 (non-NAT) > [Aug 3 09:40:28] Found user '3254101' > [Aug 3 09:40:28] Found RTP audio format 0 > [Aug 3 09:40:28] Found RTP audio format 18 > [Aug 3 09:40:28] Found RTP audio format 101 > [Aug 3 09:40:28] Peer audio RTP is at port xxx.yyy.128.18:2260 > [Aug 3 09:40:28] Found description format PCMU > [Aug 3 09:40:28] Found description format G729 > [Aug 3 09:40:28] Found description format telephone-event > [Aug 3 09:40:28] Capabilities: us - 0x104 (ulaw|g729), peer > - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - > 0x104 (ulaw|g729) > [Aug 3 09:40:28] Non-codec capabilities: us - 0x1 > (telephone-event), peer - 0x1 (telephone-event), combined - > 0x1 (telephone-event) > [Aug 3 09:40:28] Looking for 9220370 in pbx_betty_start > (domain labpbx1.ipt.twoeighty.com) > [Aug 3 09:40:28] list_route: hop: <sip:3254101@xxx.yyy.128.18> > [Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.128.18:5060: > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone> > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Content-Length: 0 > > > --- > [Aug 3 09:40:28] -- Executing NoOp("SIP/3254101-3ebc", > "*** OnNet originated call "Chocolate Chip" <3254101> -> > 9220370") in new stack > [Aug 3 09:40:28] -- Executing AGI("SIP/3254101-3ebc", > "ipt/originator.py") in new stack > [Aug 3 09:40:28] -- Launched AGI Script > /var/lib/asterisk/agi-bin/ipt/originator.py > [Aug 3 09:40:28] -- AGI Script Executing Application: > (SetAccount) Options: (9220370) > [Aug 3 09:40:28] -- AGI Script Executing Application: > (ChanIsAvail) Options: (SIP/9220370) > [Aug 3 09:40:28] Destroying call > '46b578613a629ebb69171e8b7f9b8458@xxx.yyy.142.162' > [Aug 3 09:40:28] -- AGI Script Executing Application: > (Dial) Options: (SIP/dundisip2/9220370) > [Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164 > [Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP > [Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP > [Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP > [Aug 3 09:40:28] 14 headers, 12 lines > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.142.163:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13082 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > [Aug 3 09:40:28] -- Called dundisip2/9220370 > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.142.163:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Proxy-Authenticate: Digest realm="ipt.twoeighty.com", nonce="122095ae" > Content-Length: 0 > > > [Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- > [Aug 3 09:40:28] We're at xxx.yyy.142.162 port 27164 > [Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP > [Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP > [Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.142.163:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="122095ae", > response="c3812ae6a639df3827b26ff969acad23", opaque="" > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13083 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.142.163:5060: > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Content-Length: 0 > > > [Aug 3 09:40:28] --- (10 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Transmitting (no NAT) to xxx.yyy.142.163:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- > [Aug 3 09:40:28] WARNING[13063]: chan_sip.c:9696 > handle_response_invite: Forbidden - wrong password on > authentication for INVITE to '"Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86' > [Aug 3 09:40:28] -- SIP/dundisip2-936f is circuit-busy > [Aug 3 09:40:28] == Everyone is busy/congested at this > time (1:0/1/0) > [Aug 3 09:40:28] -- AGI Script Executing Application: > (Dial) Options: (Local/u9220370@global_vmdeposit) > [Aug 3 09:40:28] -- Called u9220370@global_vmdeposit > [Aug 3 09:40:28] -- Executing > Answer("Local/u9220370@global_vmdeposit-3666,2", "") in new stack > [Aug 3 09:40:28] -- > Local/u9220370@global_vmdeposit-3666,1 answered SIP/3254101-3ebc > [Aug 3 09:40:28] -- Executing > Wait("Local/u9220370@global_vmdeposit-3666,2", "1") in new stack > [Aug 3 09:40:28] We're at xxx.yyy.142.162 port 28718 > [Aug 3 09:40:28] Adding codec 0x4 (ulaw) to SDP > [Aug 3 09:40:28] Adding codec 0x100 (g729) to SDP > [Aug 3 09:40:28] Adding non-codec 0x1 (telephone-event) to SDP > [Aug 3 09:40:28] Reliably Transmitting (no NAT) to > xxx.yyy.128.18:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bK23dd36ae56C5E669;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 2 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13082 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 28718 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- > [Aug 3 09:40:28] > <-- SIP read from xxx.yyy.128.18:5060: > ACK sip:9220370@xxx.yyy.142.162 SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKfc84778d745812F4 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > CSeq: 2 ACK > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, > SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Max-Forwards: 70 > Content-Length: 0 > > > [Aug 3 09:40:28] --- (11 headers 0 lines)[Aug 3 09:40:28] --- > [Aug 3 09:40:28] Destroying call > '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' > [Aug 3 09:40:29] == Spawn extension (pbx_betty_start, > 9220370, 2) exited non-zero on > 'Local/u9220370@global_vmdeposit-3666,2<ZOMBIE>' > [Aug 3 09:40:29] -- Executing > VoiceMail("SIP/3254101-3ebc", "u9220370@voicemail") in new stack > uniqueid => 58 > customer_id => 0 > context => voicemail > mailbox => 9220370 > password => 1234 > operator => No > attach => Yes > delete => No > stamp => 0000-00-00 00:00:00 > envelope => Yes > saycid => Yes > [Aug 3 09:40:29] -- Playing 'vm-theperson' (language 'en') > [Aug 3 09:40:30] > <-- SIP read from xxx.yyy.128.18:5060: > BYE sip:9220370@xxx.yyy.142.162 SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > CSeq: 3 BYE > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > Contact: <sip:3254101@xxx.yyy.128.18> > User-Agent: PolycomSoundPointIP-SPIP_600-UA/1.6.3.0067 > Proxy-Authorization: Digest username="3254101", > realm="ipt.twoeighty.com", nonce="673c57e6", > uri="sip:9220370@labpbx1.ipt.twoeighty.com;user=phone", > response="9d5875c8a16dfdf29ac5a43a02e44eec", algorithm=MD5 > Max-Forwards: 70 > Content-Length: 0 > > > [Aug 3 09:40:30] --- (11 headers 0 lines)[Aug 3 09:40:30] --- > [Aug 3 09:40:30] Sending to xxx.yyy.128.18 : 5060 (non-NAT) > [Aug 3 09:40:30] Transmitting (no NAT) to xxx.yyy.128.18:5060: > SIP/2.0 200 OK > Via: SIP/2.0/UDP > xxx.yyy.128.18;branch=z9hG4bKcaab7aa7CAA53996;received=xxx.yyy.128.18 > From: "Chocolate Chip" > <sip:3254101@labpbx1.ipt.twoeighty.com>;tag=6140575F-F74900CC > To: <sip:9220370@labpbx1.ipt.twoeighty.com;user=phone>;tag=as4889fec6 > Call-ID: 86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18 > CSeq: 3 BYE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.162> > Content-Length: 0 > > > --- > [Aug 3 09:40:30] == Spawn extension (global_vmdeposit, > u9220370, 3) exited non-zero on 'SIP/3254101-3ebc' > [Aug 3 09:40:31] Destroying call > '86e7a3db-bc712205-5e89a5da@xxx.yyy.128.18' > > > and here's a full sip console trace on the second pbx box. > > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13082 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- (14 headers 12 lines)--- > Using INVITE request as basis request - > 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > Sending to xxx.yyy.142.162 : 5060 (NAT) > Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060: > SIP/2.0 407 Proxy Authentication Required > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Proxy-Authenticate: Digest realm="ipt.twoeighty.com", nonce="122095ae" > Content-Length: 0 > > > --- > Scheduling destruction of call > '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' in 15000 ms > Found user '3254101' > > <-- SIP read from xxx.yyy.142.162:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK23000840;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 102 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- (11 headers 0 lines)--- > > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="122095ae", > response="c3812ae6a639df3827b26ff969acad23", opaque="" > Date: Thu, 03 Aug 2006 15:40:28 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 13082 13083 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 27164 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > > --- (15 headers 12 lines)--- > Using INVITE request as basis request - > 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > Sending to xxx.yyy.142.162 : 5060 (NAT) > Found user '3254101' > Aug 3 09:40:25 NOTICE[10884]: chan_sip.c:10469 > handle_request_invite: Failed to authenticate user "Chocolate > Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > Reliably Transmitting (no NAT) to xxx.yyy.142.162:5060: > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport;received=xxx > .yyy.142.162 > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Contact: <sip:9220370@xxx.yyy.142.163> > Content-Length: 0 > > > --- > > <-- SIP read from xxx.yyy.142.162:5060: > ACK sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK188cc77a;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as097cbc86 > To: <sip:9220370@labpbx2.ipt.twoeighty.com>;tag=as004358bd > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162 > CSeq: 103 ACK > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Content-Length: 0 > > > --- (11 headers 0 lines)--- > Destroying call '7cd3cdfc64631a6f243aed9968503ae2@xxx.yyy.142.162' > > > > > -----Original Message----- > > From: Watkins, Bradley > > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf > Of Watkins, > > Bradley > > Sent: Wednesday, August 02, 2006 6:31 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > Could you perhaps post a sip debug from both of the Asterisk > > consoles for the respective peers? > > ? > > I think we're very close, and I definitely want to get this > > working for you. > > ? > > Regards, > > - Brad > > > > _____ > > > > From: asterisk-users-bounces@lists.digium.com on behalf of > > Douglas Garstang > > Sent: Wed 8/2/2006 7:16 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > Bradley, > > > > I changed the type from friend to peer in sip.conf... > > > > [dundisip1] > > type=peer > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.twoeighty.com > > qualify=yes > > > > [dundisip2] > > type=peer > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.twoeighty.com > > qualify=yes > > > > but that just yielded the same error... > > > > [Aug? 2 17:07:51] NOTICE[10971]: chan_sip.c:9685 > > handle_response_invite: Failed to authenticate on INVITE to > > '"Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as5e6e3efe' > > > > Btw, here's a sip trace between the asterisk boxes... > > > > Capturing on eth0 > > 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > > 407 Proxy Authentication Required > > 3?? 0.177497 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@labpbx1.ipt.oneeighty.com > > 4?? 0.329036 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 5?? 0.329492 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > 100 Trying > > 6?? 0.643513 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP > > Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with > > session description > > 7?? 0.644104 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: > > 407 Proxy Authentication Required > > 8?? 0.644356 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: > > ACK sip:9220370@labpbx2.ipt.oneeighty.com > > 9?? 0.647990 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP > > Status: 200 OK, with session description > > 10?? 0.806136 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@xxx.yyy.142.162 > > 11?? 2.280664 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > BYE sip:9220370@xxx.yyy.142.162 > > 12?? 2.280780 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: 200 OK > > > > You can see that the first system doesn't resend the INVITE > > with the auth credentials as requested. > > > > So, then I also put the username field, and sip.conf now > > looks like this: > > > > [dundisip1] > > type=peer > > username=dundisip1 > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.twoeighty.com > > qualify=yes > > > > [dundisip2] > > type=peer > > username=dundisip2 > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.twoeeighty.com > > qualify=yes > > > > and with dundi.conf as: > > > > 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER} > > 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER} > > > > A CLI lookup still yields: > > > > *CLI> dundi lookup 9220370@180netsip > > ? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH) > > ???? from 00:14:22:1e:2a:d0, expires in 0 s > > DUNDi lookup completed in 130 ms > > > > HOWEVER, at attempt to dial results in this now: > > > > [Aug? 2 17:14:00] WARNING[11178]: chan_sip.c:9696 > > handle_response_invite: Forbidden - wrong password on > > authentication for INVITE to '"Chocolate Chip" > > <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f' > > > > And here's the SIP trace for THAT! > > > > 1?? 0.000000 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 2?? 0.000311 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > > 407 Proxy Authentication Required > > 3?? 0.158008 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@labpbx1.ipt.twoeighty.com > > 4?? 0.295054 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP/SDP > > Request: INVITE > > sip:9220370@labpbx1.ipt.twoeighty.com;user=phone, with > > session description > > 5?? 0.300557 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP Status: > 100 Trying > > 6?? 0.602324 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP > > Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with > > session description > > 7?? 0.603002 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: > > 407 Proxy Authentication Required > > 8?? 0.603303 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: > > ACK sip:9220370@labpbx2.ipt.twoeighty.com > > 9?? 0.603485 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP/SDP > > Request: INVITE sip:9220370@labpbx2.ipt.twoeighty.com, with > > session description > > 10?? 0.604251 xxx.yyy.142.163 -> xxx.yyy.142.162 SIP Status: > > 403 Forbidden > > 11?? 0.604553 xxx.yyy.142.162 -> xxx.yyy.142.163 SIP Request: > > ACK sip:9220370@labpbx2.ipt.twoeighty.com > > 12?? 0.608324 xxx.yyy.142.162 -> xxx.yyy.128.18 SIP/SDP > > Status: 200 OK, with session description > > 13?? 0.749668 xxx.yyy.128.18 -> xxx.yyy.142.162 SIP Request: > > ACK sip:9220370@xxx.yyy.142.162 > > > > And on the second Asterisk console we have logged: > > > > Aug? 2 17:13:57 NOTICE[29764]: chan_sip.c:10469 > > handle_request_invite: Failed to authenticate user "Chocolate > > Chip" <sip:3254101@xxx.yyy.142.162>;tag=as6090e60f > > > > Ugh... not having much luck with this... > > > > > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > > > Sent: Wednesday, August 02, 2006 5:01 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > Try putting a username= in the peer (BTW, use peer not friend) > > > definitions.? You appear to be attempting to authenticate as the > > > originating callerid (3254101). > > > > > > - Brad > > > > > > -----Original Message----- > > > From: asterisk-users-bounces@lists.digium.com > > > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf > > Of Douglas > > > Garstang > > > Sent: Wednesday, August 02, 2006 6:45 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > -----Original Message----- > > > > From: Aaron Daniel [mailto:amdtech@shsu.edu] > > > > Sent: Wednesday, August 02, 2006 4:02 PM > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > > > > > > > > > On Wed, 2006-08-02 at 15:52 -0600, Douglas Garstang wrote: > > > > > Well yes, it looked dubious to me too, although I can't > > > > find the syntaxt documented anywhere. > > > > > However, that's what DUNDis giving me as a path to the phone! > > > > > > > > > > Something is screwed with DUNDi and SIP. Has ANYONE > > > > actually implemnted it? > > > > > I can't find it documented anywhere > > > > > > > > > > Doug. > > > > > > > > DUNDi gives you only what you give it to give you.? > You're the one > > > > that needs to set the dial string correctly in DUNDi to get > > > one back > > > > that works.? DUNDi is only as automatic as you let it be. > > > > > > > > This is what ours looks like.? We don't use the iax > > > versions (mainly > > > > cause I want a homogenous SIP system), but we have entries > > > in sip.conf > > > > > > > include files for each of the servers so we just dial > > > > ${server}/${number}.? This has been working for us for > > > about 2 months > > > > now, pretty much flawlessly as long as the phone's registered. > > > > > > > > e164 => dundi-extens,0,SIP,scm1/${NUMBER} e164-iax => > > > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > > > internal => dundi-extens,0,SIP,scm1/${NUMBER} internal-iax => > > > > dundi-extens,0,IAX2,dundi:${SECRET}@scm1/${NUMBER} > > > > > > > > [scm1] > > > > type=friend > > > > secret=p4ssw0rd > > > > insecure=very > > > > context=incoming > > > > host=scm1.shsu.edu > > > > qualify=yes > > > > nat=no > > > > > > Aaron, while not really sure what I was doing, but extending > > > beyond your > > > example, I gave this a shot: > > > > > > dundi.conf: > > > 180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER} > > > 180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER} > > > > > > sip.conf: > > > [dundisip1] > > > type=friend > > > secret=password > > > insecure=very > > > context=global_dundi_local > > > host=labpbx1.ipt.twoeighty.com > > > qualify=yes > > > > > > [dundisip2] > > > type=friend > > > secret=password > > > insecure=very > > > context=global_dundi_local > > > host=labpbx2.ipt.twoeighty.com > > > qualify=yes > > > > > > A CLI lookup looks better... > > > > > > *CLI> dundi lookup 9220370@180netsip > > >?? 1.???? 0 SIP/dundisip2/9220370 (EXISTS|CANMATCH) > > >????? from 00:14:22:1e:2a:d0, expires in 0 s DUNDi lookup > > completed in > > > 107 ms > > > > > > However, my dial still fails... > > > > > > [Aug? 2 16:42:44]???? -- AGI Script Executing Application: (Dial) > > > Options: (SIP/dundisip2/9220370) > > > [Aug? 2 16:42:44]???? -- Called dundisip2/9220370 > > > [Aug? 2 16:42:44] NOTICE[10474]: chan_sip.c:9685 > > > handle_response_invite: > > > Failed to authenticate on INVITE to '"Chocolate Chip" > > > <sip:3254101@xxx.yyy.142.162>;tag=as4f13a2f1' > > > [Aug? 2 16:42:44]???? -- SIP/dundisip2-7b30 is circuit-busy > > > > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >??? http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > The contents of this e-mail are intended for the named > > > addressee only. It contains information that may be > > > confidential. Unless you are the named addressee or an > > > authorized designee, you may not copy or use it, or disclose > > > it to anyone else. If you received it in error please notify > > > us immediately and then destroy it. > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > >??? http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > ?? http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > The contents of this e-mail are intended for the named > > addressee only. It contains information that may be > > confidential. Unless you are the named addressee or an > > authorized designee, you may not copy or use it, or disclose > > it to anyone else. If you received it in error please notify > > us immediately and then destroy it. > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > The contents of this e-mail are intended for the named > addressee only. It contains information that may be > confidential. Unless you are the named addressee or an > authorized designee, you may not copy or use it, or disclose > it to anyone else. If you received it in error please notify > us immediately and then destroy it. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Brad... Here's the INVITE that the second asterisk box receives from the first Asterisk box, after the second asterisk box sends a Proxy auth message to the first. The first sends the dundisip userid, but for some reason the second asterisk box is matching it against the From: 3254101 address. Do I need more than just the two sip.conf entries, dundisip, and dundisip2 as type=peer? Anyone? Doug.> > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as34445cb0 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="012b66e9", > response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque="" > Date: Thu, 03 Aug 2006 19:52:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 16558 16559 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 28454 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - -
I forget if this does what you want, but try adding a fromuser setting to yoyr peer entries. - Brad -----Original Message----- From: Douglas Garstang [mailto:dgarstang@oneeighty.com] Sent: Thu Aug 03 15:58:50 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: DUNDi with SIP Brad... Here's the INVITE that the second asterisk box receives from the first Asterisk box, after the second asterisk box sends a Proxy auth message to the first. The first sends the dundisip userid, but for some reason the second asterisk box is matching it against the From: 3254101 address. Do I need more than just the two sip.conf entries, dundisip, and dundisip2 as type=peer? Anyone? Doug.> > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as34445cb0 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="012b66e9", > response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque="" > Date: Thu, 03 Aug 2006 19:52:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 16558 16559 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 28454 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - -_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060803/38697216/attachment.htm
I don't think that will do it. SInce I started with Asterisk a year ago, I've wrestled the ENTIRE time with the sip conf file. It really makes no sense to me after doing this 8-5 for 10 months now. The first system is sending the correct user id. Do I need two entries in dundi.conf? Do I need two or four (in and out) users in sip.conf? Should they be peers or friends? :( -----Original Message----- From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] Sent: Thursday, August 03, 2006 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: DUNDi with SIP I forget if this does what you want, but try adding a fromuser setting to yoyr peer entries. - Brad -----Original Message----- From: Douglas Garstang [ mailto:dgarstang@oneeighty.com] Sent: Thu Aug 03 15:58:50 2006 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: DUNDi with SIP Brad... Here's the INVITE that the second asterisk box receives from the first Asterisk box, after the second asterisk box sends a Proxy auth message to the first. The first sends the dundisip userid, but for some reason the second asterisk box is matching it against the From: 3254101 address. Do I need more than just the two sip.conf entries, dundisip, and dundisip2 as type=peer? Anyone? Doug.> > <-- SIP read from xxx.yyy.142.162:5060: > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport > From: "Chocolate Chip" <sip:3254101@xxx.yyy.142.162>;tag=as34445cb0 > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > Contact: <sip:3254101@xxx.yyy.142.162> > Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162 > CSeq: 103 INVITE > User-Agent: Asterisk PBX > Max-Forwards: 70 > Remote-Party-ID: "Chocolate Chip" > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > Proxy-Authorization: Digest username="dundisip2", > realm="ipt.twoeighty.com", algorithm=MD5, > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > nonce="012b66e9", > response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque="" > Date: Thu, 03 Aug 2006 19:52:26 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Content-Type: application/sdp > Content-Length: 269 > > v=0 > o=root 16558 16559 IN IP4 xxx.yyy.142.162 > s=session > c=IN IP4 xxx.yyy.142.162 > t=0 0 > m=audio 28454 RTP/AVP 0 18 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - -_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060803/41910c2c/attachment.htm
*scream* I just got it to work with this...
180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER},nopartial
180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER},nopartial
[dundisip1]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.oneeighty.com
qualify=no
username=dundisip1
fromuser=dundisip1
[dundisip2]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.oneeighty.com
qualify=no
username=dundisip2
fromuser=dundisip2
It works... but the first system sends dundisip2 as the authid, not dundisip1. I
assume that's because it matches the host labpbx2. Well, it should be
sending dundisip1.... how can I get it to do that?
Doug.
-----Original Message-----
From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
Sent: Thursday, August 03, 2006 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
I forget if this does what you want, but try adding a fromuser setting to yoyr
peer entries.
- Brad
-----Original Message-----
From: Douglas Garstang [ mailto:dgarstang@oneeighty.com]
Sent: Thu Aug 03 15:58:50 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Brad...
Here's the INVITE that the second asterisk box receives from
the first Asterisk box, after the second asterisk box sends a
Proxy auth message to the first. The first sends the dundisip
userid, but for some reason the second asterisk box is
matching it against the From: 3254101 address. Do I need more
than just the two sip.conf entries, dundisip, and dundisip2
as type=peer?
Anyone?
Doug.
>
> <-- SIP read from xxx.yyy.142.162:5060:
> INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
> Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport
> From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as34445cb0
> To: <sip:9220370@labpbx2.ipt.twoeighty.com>
> Contact: <sip:3254101@xxx.yyy.142.162>
> Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Remote-Party-ID: "Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
> Proxy-Authorization: Digest username="dundisip2",
> realm="ipt.twoeighty.com", algorithm=MD5,
> uri="sip:9220370@labpbx2.ipt.twoeighty.com",
> nonce="012b66e9",
> response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque=""
> Date: Thu, 03 Aug 2006 19:52:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 269
>
> v=0
> o=root 16558 16559 IN IP4 xxx.yyy.142.162
> s=session
> c=IN IP4 xxx.yyy.142.162
> t=0 0
> m=audio 28454 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
_______________________________________________
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Easy, you change the fromuser=dundisip2 line to fromuser=dundisip1.
You're just matching the destination peer (dunsip2,
labpbx2.ipt.oneeighty.com) and sending the configured fromuser.
- Brad
________________________________
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas
Garstang
Sent: Thursday, August 03, 2006 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
*scream* I just got it to work with this...
180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER},nopartial
180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER},nopartial
[dundisip1]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.oneeighty.com
qualify=no
username=dundisip1
fromuser=dundisip1
[dundisip2]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.oneeighty.com
qualify=no
username=dundisip2
fromuser=dundisip2
It works... but the first system sends dundisip2 as the authid, not
dundisip1. I assume that's because it matches the host labpbx2. Well, it
should be sending dundisip1.... how can I get it to do that?
Doug.
-----Original Message-----
From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
Sent: Thursday, August 03, 2006 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
I forget if this does what you want, but try adding a fromuser
setting to yoyr peer entries.
- Brad
-----Original Message-----
From: Douglas Garstang [mailto:dgarstang@oneeighty.com]
Sent: Thu Aug 03 15:58:50 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Brad...
Here's the INVITE that the second asterisk box receives from
the first Asterisk box, after the second asterisk box sends a
Proxy auth message to the first. The first sends the dundisip
userid, but for some reason the second asterisk box is
matching it against the From: 3254101 address. Do I need more
than just the two sip.conf entries, dundisip, and dundisip2
as type=peer?
Anyone?
Doug.
>
> <-- SIP read from xxx.yyy.142.162:5060:
> INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
> Via: SIP/2.0/UDP
xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport
> From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as34445cb0
> To: <sip:9220370@labpbx2.ipt.twoeighty.com>
> Contact: <sip:3254101@xxx.yyy.142.162>
> Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Remote-Party-ID: "Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
> Proxy-Authorization: Digest username="dundisip2",
> realm="ipt.twoeighty.com", algorithm=MD5,
> uri="sip:9220370@labpbx2.ipt.twoeighty.com",
> nonce="012b66e9",
> response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque=""
> Date: Thu, 03 Aug 2006 19:52:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
NOTIFY
> Content-Type: application/sdp
> Content-Length: 269
>
> v=0
> o=root 16558 16559 IN IP4 xxx.yyy.142.162
> s=session
> c=IN IP4 xxx.yyy.142.162
> t=0 0
> m=audio 28454 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
The contents of this e-mail are intended for the named addressee
only. It contains information that may be confidential. Unless you are
the named addressee or an authorized designee, you may not copy or use
it, or disclose it to anyone else. If you received it in error please
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Didn't work Brad. I changed fromuser to dundisip1 from dundisip2. The first
Asterisk box sends dundisip1.
[dundisip1]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.oneeighty.com
qualify=no
username=dundisip1
fromuser=dundisip1
[dundisip2]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.oneeighty.com
qualify=no
username=dundisip2
fromuser=dundisip1
My sip.conf now has:
but at the other end, I get:
Aug 4 08:33:25 NOTICE[20569]: chan_sip.c:10469 handle_request_invite: Failed to
authenticate user "Chocolate Chip"
<sip:dundisip1@xxx.yyy.142.162>;tag=as1345d31c
I assume because the call is coming in from [dundisip2]. It then looks for a
username of 'dundisip2' but it's receiving dundisip1.
No wonder the sip.conf and iax.conf files drive me nuts! You spend all your time
going in circles!
Doug.
-----Original Message-----
From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
Sent: Friday, August 04, 2006 4:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Easy, you change the fromuser=dundisip2 line to fromuser=dundisip1. You're
just matching the destination peer (dunsip2, labpbx2.ipt.oneeighty.com) and
sending the configured fromuser.
- Brad
_____
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Douglas Garstang
Sent: Thursday, August 03, 2006 4:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
*scream* I just got it to work with this...
180netsip => global_dundi_local,0,SIP,dundisip1/${NUMBER},nopartial
180netsip => global_dundi_local,0,SIP,dundisip2/${NUMBER},nopartial
[dundisip1]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx1.ipt.oneeighty.com
qualify=no
username=dundisip1
fromuser=dundisip1
[dundisip2]
type=friend
secret=password
insecure=very
context=global_dundi_local
host=labpbx2.ipt.oneeighty.com
qualify=no
username=dundisip2
fromuser=dundisip2
It works... but the first system sends dundisip2 as the authid, not dundisip1. I
assume that's because it matches the host labpbx2. Well, it should be
sending dundisip1.... how can I get it to do that?
Doug.
-----Original Message-----
From: Watkins, Bradley [mailto:Bradley.Watkins@compuware.com]
Sent: Thursday, August 03, 2006 2:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
I forget if this does what you want, but try adding a fromuser setting to yoyr
peer entries.
- Brad
-----Original Message-----
From: Douglas Garstang [ mailto:dgarstang@oneeighty.com]
Sent: Thu Aug 03 15:58:50 2006
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Re: DUNDi with SIP
Brad...
Here's the INVITE that the second asterisk box receives from
the first Asterisk box, after the second asterisk box sends a
Proxy auth message to the first. The first sends the dundisip
userid, but for some reason the second asterisk box is
matching it against the From: 3254101 address. Do I need more
than just the two sip.conf entries, dundisip, and dundisip2
as type=peer?
Anyone?
Doug.
>
> <-- SIP read from xxx.yyy.142.162:5060:
> INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0
> Via: SIP/2.0/UDP xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport
> From: "Chocolate Chip"
<sip:3254101@xxx.yyy.142.162>;tag=as34445cb0
> To: <sip:9220370@labpbx2.ipt.twoeighty.com>
> Contact: <sip:3254101@xxx.yyy.142.162>
> Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162
> CSeq: 103 INVITE
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Remote-Party-ID: "Chocolate Chip"
> <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no
> Proxy-Authorization: Digest username="dundisip2",
> realm="ipt.twoeighty.com", algorithm=MD5,
> uri="sip:9220370@labpbx2.ipt.twoeighty.com",
> nonce="012b66e9",
> response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque=""
> Date: Thu, 03 Aug 2006 19:52:26 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Content-Type: application/sdp
> Content-Length: 269
>
> v=0
> o=root 16558 16559 IN IP4 xxx.yyy.142.162
> s=session
> c=IN IP4 xxx.yyy.142.162
> t=0 0
> m=audio 28454 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
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Steve, I don't see how it could possibly work. The second system sees a connection from dundisip1 on labpbx2, so it then looks at the dundisip2 entry because it's matching against the host. It then looks at the username, but it's dundisip2... which doesn't match dundisip1, and it fails the call.> -----Original Message----- > From: Steve Totaro [mailto:stotaro@asteriskhelpdesk.com] > Sent: Friday, August 04, 2006 8:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: DUNDi with SIP > > > Works fine for me. > > Douglas Garstang wrote: > > Didn't work Brad. I changed fromuser to dundisip1 from > dundisip2. The > > first Asterisk box sends dundisip1. > > > > [dundisip1] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.oneeighty.com > > qualify=no > > username=dundisip1 > > fromuser=dundisip1 > > > > [dundisip2] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.oneeighty.com > > qualify=no > > username=dundisip2 > > fromuser=dundisip1 > > > > > > My sip.conf now has: > > > > but at the other end, I get: > > > > Aug 4 08:33:25 NOTICE[20569]: chan_sip.c:10469 > handle_request_invite: > > Failed to authenticate user "Chocolate Chip" > > <sip:dundisip1@xxx.yyy.142.162>;tag=as1345d31c > > > > I assume because the call is coming in from [dundisip2]. It > then looks > > for a username of 'dundisip2' but it's receiving dundisip1. > > > > No wonder the sip.conf and iax.conf files drive me nuts! > You spend all > > your time going in circles! > > > > Doug. > > > > > > > > -----Original Message----- > > *From:* Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > > *Sent:* Friday, August 04, 2006 4:20 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* RE: [asterisk-users] Re: DUNDi with SIP > > > > Easy, you change the fromuser=dundisip2 line to > > fromuser=dundisip1. You're just matching the destination peer > > (dunsip2, labpbx2.ipt.oneeighty.com) and sending the configured > > fromuser. > > > > - Brad > > > > > -------------------------------------------------------------- > ---------- > > *From:* asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] *On Behalf Of > > *Douglas Garstang > > *Sent:* Thursday, August 03, 2006 4:18 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* RE: [asterisk-users] Re: DUNDi with SIP > > > > *scream* I just got it to work with this... > > > > 180netsip => > global_dundi_local,0,SIP,dundisip1/${NUMBER},nopartial > > 180netsip => > global_dundi_local,0,SIP,dundisip2/${NUMBER},nopartial > > > > [dundisip1] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.oneeighty.com > > qualify=no > > username=dundisip1 > > fromuser=dundisip1 > > > > [dundisip2] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.oneeighty.com > > qualify=no > > username=dundisip2 > > fromuser=dundisip2 > > It works... but the first system sends dundisip2 as the authid, > > not dundisip1. I assume that's because it matches the host > > labpbx2. Well, it should be sending dundisip1.... how > can I get it > > to do that? > > > > Doug. > > > > -----Original Message----- > > *From:* Watkins, Bradley > [mailto:Bradley.Watkins@compuware.com] > > *Sent:* Thursday, August 03, 2006 2:06 PM > > *To:* Asterisk Users Mailing List - Non-Commercial > Discussion > > *Subject:* RE: [asterisk-users] Re: DUNDi with SIP > > > > I forget if this does what you want, but try adding > a fromuser > > setting to yoyr peer entries. > > > > - Brad > > > > -----Original Message----- > > From: Douglas Garstang [mailto:dgarstang@oneeighty.com] > > Sent: Thu Aug 03 15:58:50 2006 > > To: Asterisk Users Mailing List - > Non-Commercial Discussion > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > Brad... > > > > Here's the INVITE that the second asterisk box receives from > > the first Asterisk box, after the second asterisk > box sends a > > Proxy auth message to the first. The first sends > the dundisip > > userid, but for some reason the second asterisk box is > > matching it against the From: 3254101 address. Do I > need more > > than just the two sip.conf entries, dundisip, and dundisip2 > > as type=peer? > > > > Anyone? > > > > Doug. > > > > > > > > <-- SIP read from xxx.yyy.142.162:5060: > > > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > > > Via: SIP/2.0/UDP > > xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport > > > From: "Chocolate Chip" > > <sip:3254101@xxx.yyy.142.162>;tag=as34445cb0 > > > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > > > Contact: <sip:3254101@xxx.yyy.142.162> > > > Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162 > > > CSeq: 103 INVITE > > > User-Agent: Asterisk PBX > > > Max-Forwards: 70 > > > Remote-Party-ID: "Chocolate Chip" > > > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > > > Proxy-Authorization: Digest username="dundisip2", > > > realm="ipt.twoeighty.com", algorithm=MD5, > > > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > > > nonce="012b66e9", > > > response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque="" > > > Date: Thu, 03 Aug 2006 19:52:26 GMT > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, > > NOTIFY > > > Content-Type: application/sdp > > > Content-Length: 269 > > > > > > v=0 > > > o=root 16558 16559 IN IP4 xxx.yyy.142.162 > > > s=session > > > c=IN IP4 xxx.yyy.142.162 > > > t=0 0 > > > m=audio 28454 RTP/AVP 0 18 101 > > > a=rtpmap:0 PCMU/8000 > > > a=rtpmap:18 G729/8000 > > > a=fmtp:18 annexb=no > > > a=rtpmap:101 telephone-event/8000 > > > a=fmtp:101 0-16 > > > a=silenceSupp:off - - - - > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > The contents of this e-mail are intended for the named > > addressee only. It contains information that may be > > confidential. Unless you are the named addressee or an > > authorized designee, you may not copy or use it, or disclose > > it to anyone else. If you received it in error please notify > > us immediately and then destroy it. > > > > The contents of this e-mail are intended for the named addressee > > only. It contains information that may be confidential. > Unless you > > are the named addressee or an authorized designee, you may not > > copy or use it, or disclose it to anyone else. If you > received it > > in error please notify us immediately and then destroy it. > > > > > -------------------------------------------------------------- > ---------- > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
Well, after 24 hours I'm still trying to get SIP trunking to work. I'm going in circles and it's driving me nuts. Here's my latest sip.conf, that I have on both systems. system 1 sends dundisip1_in as the userid to system 2. System 2 asks for proxy auth. System 1 sends an ACK and doesn't do anything else. It then logs on the console: [Aug 4 10:19:59] NOTICE[30136]: chan_sip.c:9685 handle_response_invite: Failed to authenticate on INVITE to '"Chocolate Chip" <sip:dundisip1_in@xxx.yyy.142.162>;tag=as7da2967e' :( [dundisip1] type=peer secret=password insecure=very host=labpbx1.ipt.twoeighty.com qualify=no fromuser=dundisip1_in [dundisip2] type=peer secret=password insecure=very host=labpbx2.ipt.twoeighty.com qualify=no fromuser=dundisip2_in [dundisip1_in] type=user username=dundisip1_in secret=password insecure=very context=global_dundi_local host=labpbx2.ipt.twoeighty.com qualify=no [dundisip2_in] type=user username=dundisip2_in secret=password insecure=very context=global_dundi_local host=labpbx1.ipt.twoeighty.com qualify=no> -----Original Message----- > From: Steve Totaro [mailto:stotaro@asteriskhelpdesk.com] > Sent: Friday, August 04, 2006 8:51 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Re: DUNDi with SIP > > > Works fine for me. > > Douglas Garstang wrote: > > Didn't work Brad. I changed fromuser to dundisip1 from > dundisip2. The > > first Asterisk box sends dundisip1. > > > > [dundisip1] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.oneeighty.com > > qualify=no > > username=dundisip1 > > fromuser=dundisip1 > > > > [dundisip2] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.oneeighty.com > > qualify=no > > username=dundisip2 > > fromuser=dundisip1 > > > > > > My sip.conf now has: > > > > but at the other end, I get: > > > > Aug 4 08:33:25 NOTICE[20569]: chan_sip.c:10469 > handle_request_invite: > > Failed to authenticate user "Chocolate Chip" > > <sip:dundisip1@xxx.yyy.142.162>;tag=as1345d31c > > > > I assume because the call is coming in from [dundisip2]. It > then looks > > for a username of 'dundisip2' but it's receiving dundisip1. > > > > No wonder the sip.conf and iax.conf files drive me nuts! > You spend all > > your time going in circles! > > > > Doug. > > > > > > > > -----Original Message----- > > *From:* Watkins, Bradley [mailto:Bradley.Watkins@compuware.com] > > *Sent:* Friday, August 04, 2006 4:20 AM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* RE: [asterisk-users] Re: DUNDi with SIP > > > > Easy, you change the fromuser=dundisip2 line to > > fromuser=dundisip1. You're just matching the destination peer > > (dunsip2, labpbx2.ipt.oneeighty.com) and sending the configured > > fromuser. > > > > - Brad > > > > > -------------------------------------------------------------- > ---------- > > *From:* asterisk-users-bounces@lists.digium.com > > [mailto:asterisk-users-bounces@lists.digium.com] *On Behalf Of > > *Douglas Garstang > > *Sent:* Thursday, August 03, 2006 4:18 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > > *Subject:* RE: [asterisk-users] Re: DUNDi with SIP > > > > *scream* I just got it to work with this... > > > > 180netsip => > global_dundi_local,0,SIP,dundisip1/${NUMBER},nopartial > > 180netsip => > global_dundi_local,0,SIP,dundisip2/${NUMBER},nopartial > > > > [dundisip1] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx1.ipt.oneeighty.com > > qualify=no > > username=dundisip1 > > fromuser=dundisip1 > > > > [dundisip2] > > type=friend > > secret=password > > insecure=very > > context=global_dundi_local > > host=labpbx2.ipt.oneeighty.com > > qualify=no > > username=dundisip2 > > fromuser=dundisip2 > > It works... but the first system sends dundisip2 as the authid, > > not dundisip1. I assume that's because it matches the host > > labpbx2. Well, it should be sending dundisip1.... how > can I get it > > to do that? > > > > Doug. > > > > -----Original Message----- > > *From:* Watkins, Bradley > [mailto:Bradley.Watkins@compuware.com] > > *Sent:* Thursday, August 03, 2006 2:06 PM > > *To:* Asterisk Users Mailing List - Non-Commercial > Discussion > > *Subject:* RE: [asterisk-users] Re: DUNDi with SIP > > > > I forget if this does what you want, but try adding > a fromuser > > setting to yoyr peer entries. > > > > - Brad > > > > -----Original Message----- > > From: Douglas Garstang [mailto:dgarstang@oneeighty.com] > > Sent: Thu Aug 03 15:58:50 2006 > > To: Asterisk Users Mailing List - > Non-Commercial Discussion > > Subject: RE: [asterisk-users] Re: DUNDi with SIP > > > > Brad... > > > > Here's the INVITE that the second asterisk box receives from > > the first Asterisk box, after the second asterisk > box sends a > > Proxy auth message to the first. The first sends > the dundisip > > userid, but for some reason the second asterisk box is > > matching it against the From: 3254101 address. Do I > need more > > than just the two sip.conf entries, dundisip, and dundisip2 > > as type=peer? > > > > Anyone? > > > > Doug. > > > > > > > > <-- SIP read from xxx.yyy.142.162:5060: > > > INVITE sip:9220370@labpbx2.ipt.twoeighty.com SIP/2.0 > > > Via: SIP/2.0/UDP > > xxx.yyy.142.162:5060;branch=z9hG4bK047028fa;rport > > > From: "Chocolate Chip" > > <sip:3254101@xxx.yyy.142.162>;tag=as34445cb0 > > > To: <sip:9220370@labpbx2.ipt.twoeighty.com> > > > Contact: <sip:3254101@xxx.yyy.142.162> > > > Call-ID: 7ca53bec20139d6a305b54936921b0ab@xxx.yyy.142.162 > > > CSeq: 103 INVITE > > > User-Agent: Asterisk PBX > > > Max-Forwards: 70 > > > Remote-Party-ID: "Chocolate Chip" > > > <sip:3254101@xxx.yyy.142.162>;privacy=off;screen=no > > > Proxy-Authorization: Digest username="dundisip2", > > > realm="ipt.twoeighty.com", algorithm=MD5, > > > uri="sip:9220370@labpbx2.ipt.twoeighty.com", > > > nonce="012b66e9", > > > response="bd5b3881afd0bdb827915a9eeb97dcd8", opaque="" > > > Date: Thu, 03 Aug 2006 19:52:26 GMT > > > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, > SUBSCRIBE, > > NOTIFY > > > Content-Type: application/sdp > > > Content-Length: 269 > > > > > > v=0 > > > o=root 16558 16559 IN IP4 xxx.yyy.142.162 > > > s=session > > > c=IN IP4 xxx.yyy.142.162 > > > t=0 0 > > > m=audio 28454 RTP/AVP 0 18 101 > > > a=rtpmap:0 PCMU/8000 > > > a=rtpmap:18 G729/8000 > > > a=fmtp:18 annexb=no > > > a=rtpmap:101 telephone-event/8000 > > > a=fmtp:101 0-16 > > > a=silenceSupp:off - - - - > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > The contents of this e-mail are intended for the named > > addressee only. It contains information that may be > > confidential. Unless you are the named addressee or an > > authorized designee, you may not copy or use it, or disclose > > it to anyone else. If you received it in error please notify > > us immediately and then destroy it. > > > > The contents of this e-mail are intended for the named addressee > > only. It contains information that may be confidential. > Unless you > > are the named addressee or an authorized designee, you may not > > copy or use it, or disclose it to anyone else. If you > received it > > in error please notify us immediately and then destroy it. > > > > > -------------------------------------------------------------- > ---------- > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >