I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make sense any longer. Has anyone got a working example they could supply? Can I do all this with just three peers and one username? Thanks... Doug. [pbx1_inbound] type=user auth=rsa inkeys=pbx1 username=pbx1_inbound deny=0.0.0.0 permit=xxx.187.142.203 context=global_pbx_transfer [pbx1_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx1.ipt.yyy.com [pbx2_inbound] type=user auth=rsa inkeys=pbx2 username=pbx2_inbound deny=0.0.0.0 permit=xxx.187.142.204 context=global_pbx_transfer [pbx2_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx2.ipt.yyy.com [pbx3_inbound] type=user auth=rsa inkeys=pbx3 username=pbx3_inbound deny=0.0.0.0 permit=xxx.187.142.234 context=global_pbx_transfer [pbx3_outbound] type=peer auth=rsa outkey=pbx1 username=pbx3 host=pbx3.ipt.yyy.com -----Original Message----- From: George Vagenas [mailto:gvagasterisk@gmail.com] Sent: Fri 3/24/2006 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: George Vagenas Subject: Re: [Asterisk-Users] SIP trunk problem Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. On 3/25/06, Martin Joseph <ast@stillnewt.org> wrote: On Mar 24, 2006, at 1:19 PM, George Vagenas wrote: > Hi all, > > I have the following problem, working with a SIP provider, if i setup > my SJPhone to register directly to their STUN server and working over > a 384/128 ADSL i have a really good quality, but then if i configure > Asterisk to register to the same provider over the same 384/128 > circuit the quality is REALLY BAD. The obvious difference is that > using directly the SJPhone i am using STUN, while when i am using > Asterisk to connect to my SIP provider and the SJPhone to connect to > Asterisk i have the following configuration for Asterisk. > > > register => user:pass@sip.provider.com > > [mysip] > host=sip.provider.com > type=peer > qualify=yes > username=user > secret=pass > nat=yes > disallow=all > allow=ulaw > > > I am using Asterisk 1.2.3. > > I think that i am missing something or misconfigure something because > for sure its not matter of the ADSL since in both tests i am doing i > am using the same circuit. > > Any idea please???? I don't think using ulaw on a 128K bit upstream circuit is a good choice. I would use g729. Marty PS I can't be the stun server if asterisk is working, but quality is poor. _______________________________________________ --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'vce got three Asterisk systems here that I'd like to be able to place
calls between with IAX. Each system, pbx1, pbx2, pbx3, should be able to
connect to every other. Do I need separate user/peers or can I combine them
into a single user=friend for each system?
Has anyone got a working example they could supply? Can I do all this with
just three peers and one username?
Thanks... Doug.
[pbx1_inbound]
type=user
auth=rsa
inkeys=pbx1
username=pbx1_inbound
deny=0.0.0.0
permit=xxx.187.142.203
context=global_pbx_transfer
[pbx1_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx1.ipt.yyy.com
[pbx2_inbound]
type=user
auth=rsa
inkeys=pbx2
username=pbx2_inbound
deny=0.0.0.0
permit=xxx.187.142.204
context=global_pbx_transfer
[pbx2_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx2.ipt.yyy.com
[pbx3_inbound]
type=user
auth=rsa
inkeys=pbx3
username=pbx3_inbound
deny=0.0.0.0
permit=xxx.187.142.234
context=global_pbx_transfer
**********Hi Doug,
You certainly can have in/out trunks for each and work fine plus you can get
more granular with security.
If all these boxes are on the same subnet and secure from a public network,
there is really no reason to have an IAX [context] for each outbound/inbound
session between the servers, just use one [context] common on all servers.
Example iax.conf all PBX's
[iaxtrunk] (my internal iax trunk)
type=friend
auth=md5
secret=1234
host=dynamic
context=incomingiax
disallow=all
allow=ulaw
trunk=yes
extensions.conf all PBX's
[incomingiax]
Include => local (or whatever contexts the incoming iax trunks need access
to)
Now routing call between them is a whole other topic, several ways to
accomplish this but it is all dial plan related at this point. The only
thing we accomplished so far is allowing all 3 PBX's trunk access to each
other over a common [context] group. This is good, as you add PBX4, PBX5,
you just add this common [context] in iax.conf in the new servers without
the need of updating pbx1,2&3.
Routing example extensions.conf
[internal]
;To reach internal extensions on pbx1 (put this in pbx 2&3)
Exten => 1XXX,1,Dial(IAX2/iaxtrunk:1234@XXX.187.142.203/${EXTEN})
;To reach internal extensions on pbx2 (put this in pbx 1&3)
Exten => 2XXX,1,Dial(IAX2/iaxtrunk:1234@XXX.187.142.204/${EXTEN})
;To reach internal extensions on pbx3 (put this in pbx 1&2)
Exten => 3XXX,1,Dial(IAX2/iaxtrunk:1234@XXX.187.142.234/${EXTEN})
You could also specify each PBX in the [globals] context
Example
[globals]
TRUNKPBX1 => IAX2/iaxtrunk:1234@XXX.187.142.203
TRUNKPBX2 => IAX2/iaxtrunk:1234@XXX.187.142.204
TRUNKPBX3 => IAX2/iaxtrunk:1234@XXX.187.142.234
So your routing extension would look like this:
Exten => 1XXX,1,Dial(${TRUNKPBX1}/${EXTEN})
Exten => 2XXX,1,Dial(${TRUNKPBX2}/${EXTEN})
Exten => 3XXX,1,Dial(${TRUNKPBX3}/${EXTEN})
Hope this helps.
JR
12 hours later... still playing with this. Anyone got any ideas? Doug. -----Original Message----- From: Douglas Garstang Sent: Friday, March 24, 2006 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: IAX Incoming/Outgoing I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make sense any longer. Has anyone got a working example they could supply? Can I do all this with just three peers and one username? Thanks... Doug. [pbx1_inbound] type=user auth=rsa inkeys=pbx1 username=pbx1_inbound deny=0.0.0.0 permit=xxx.187.142.203 context=global_pbx_transfer [pbx1_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx1.ipt.yyy.com [pbx2_inbound] type=user auth=rsa inkeys=pbx2 username=pbx2_inbound deny=0.0.0.0 permit=xxx.187.142.204 context=global_pbx_transfer [pbx2_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx2.ipt.yyy.com [pbx3_inbound] type=user auth=rsa inkeys=pbx3 username=pbx3_inbound deny=0.0.0.0 permit=xxx.187.142.234 context=global_pbx_transfer [pbx3_outbound] type=peer auth=rsa outkey=pbx1 username=pbx3 host=pbx3.ipt.yyy.com -----Original Message----- From: George Vagenas [mailto:gvagasterisk@gmail.com] Sent: Fri 3/24/2006 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: George Vagenas Subject: Re: [Asterisk-Users] SIP trunk problem Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. On 3/25/06, Martin Joseph < ast@stillnewt.org> wrote: On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:> Hi all, > > I have the following problem, working with a SIP provider, if i setup > my SJPhone to register directly to their STUN server and working over > a 384/128 ADSL i have a really good quality, but then if i configure > Asterisk to register to the same provider over the same 384/128 > circuit the quality is REALLY BAD. The obvious difference is that > using directly the SJPhone i am using STUN, while when i am using > Asterisk to connect to my SIP provider and the SJPhone to connect to > Asterisk i have the following configuration for Asterisk. > > > register => user:pass@sip.provider.com > > [mysip] > host= sip.provider.com > type=peer > qualify=yes > username=user > secret=pass > nat=yes > disallow=all > allow=ulaw > > > I am using Asterisk 1.2.3. > > I think that i am missing something or misconfigure something because > for sure its not matter of the ADSL since in both tests i am doing i > am using the same circuit. > > Any idea please????I don't think using ulaw on a 128K bit upstream circuit is a good choice. I would use g729. Marty PS I can't be the stun server if asterisk is working, but quality is poor. _______________________________________________ --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> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060325/f9c27bc6/attachment.htm
Well, right now I have this on box1: [pbx1] type=friend auth=rsa inkeys=pbx1 outkey=pbx1 context=global_pbx_transfer host=pbx1.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.203 [pbx2] type=friend auth=rsa inkeys=pbx2 outkey=pbx1 context=global_pbx_transfer host=pbx2.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.233 [pbx3] type=friend auth=rsa inkeys=pbx3 outkey=pbx1 context=global_pbx_transfer host=pbx3.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.234 and this on box2: [pbx1] type=friend auth=rsa inkeys=pbx1 outkey=pbx2 context=global_pbx_transfer host=pbx1.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.203 [pbx2] type=friend auth=rsa inkeys=pbx2 outkey=pbx2 context=global_pbx_transfer host=pbx2.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.233 [pbx3] type=friend auth=rsa inkeys=pbx3 outkey=pbx2 context=global_pbx_transfer host=pbx3.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.234 And for some reason the calling system is sending pbx3 as the username.... Why would it do that? -----Original Message----- From: Steve Totaro [mailto:stotaro@asteriskhelpdesk.com] Sent: Saturday, March 25, 2006 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing No, you need to use different names. You can use friend rather than having separate entries for in/out. What do you get when you type iax2 show peers? You should be able to use friend and the same three entries on each box with the exception of changing the IP addresses. _____ From: Douglas Garstang [mailto:dgarstang@oneeighty.com] Sent: Saturday, March 25, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RE: IAX Incoming/Outgoing 12 hours later... still playing with this. Anyone got any ideas? Doug. -----Original Message----- From: Douglas Garstang Sent: Friday, March 24, 2006 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: IAX Incoming/Outgoing I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make sense any longer. Has anyone got a working example they could supply? Can I do all this with just three peers and one username? Thanks... Doug. [pbx1_inbound] type=user auth=rsa inkeys=pbx1 username=pbx1_inbound deny=0.0.0.0 permit=xxx.187.142.203 context=global_pbx_transfer [pbx1_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx1.ipt.yyy.com [pbx2_inbound] type=user auth=rsa inkeys=pbx2 username=pbx2_inbound deny=0.0.0.0 permit=xxx.187.142.204 context=global_pbx_transfer [pbx2_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx2.ipt.yyy.com [pbx3_inbound] type=user auth=rsa inkeys=pbx3 username=pbx3_inbound deny=0.0.0.0 permit=xxx.187.142.234 context=global_pbx_transfer [pbx3_outbound] type=peer auth=rsa outkey=pbx1 username=pbx3 host=pbx3.ipt.yyy.com -----Original Message----- From: George Vagenas [mailto:gvagasterisk@gmail.com] Sent: Fri 3/24/2006 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: George Vagenas Subject: Re: [Asterisk-Users] SIP trunk problem Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. On 3/25/06, Martin Joseph < ast@stillnewt.org> wrote: On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:> Hi all, > > I have the following problem, working with a SIP provider, if i setup > my SJPhone to register directly to their STUN server and working over > a 384/128 ADSL i have a really good quality, but then if i configure > Asterisk to register to the same provider over the same 384/128 > circuit the quality is REALLY BAD. The obvious difference is that > using directly the SJPhone i am using STUN, while when i am using > Asterisk to connect to my SIP provider and the SJPhone to connect to > Asterisk i have the following configuration for Asterisk. > > > register => user:pass@sip.provider.com > > [mysip] > host= sip.provider.com > type=peer > qualify=yes > username=user > secret=pass > nat=yes > disallow=all > allow=ulaw > > > I am using Asterisk 1.2.3. > > I think that i am missing something or misconfigure something because > for sure its not matter of the ADSL since in both tests i am doing i > am using the same circuit. > > Any idea please????I don't think using ulaw on a 128K bit upstream circuit is a good choice. I would use g729. Marty PS I can't be the stun server if asterisk is working, but quality is poor. _______________________________________________ --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> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060325/3de075a8/attachment.htm
Actually, I commented out [pbx3] on the caller, and the callee is STILL seeing pbx3 as the username. That's even more bizarre. I am sendng pbx1 as the key from the pbx1 system, and pbx2 is matching it against pbx3??? Huh??? -----Original Message----- From: Douglas Garstang Sent: Saturday, March 25, 2006 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing Well, right now I have this on box1: [pbx1] type=friend auth=rsa inkeys=pbx1 outkey=pbx1 context=global_pbx_transfer host=pbx1.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.203 [pbx2] type=friend auth=rsa inkeys=pbx2 outkey=pbx1 context=global_pbx_transfer host=pbx2.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.233 [pbx3] type=friend auth=rsa inkeys=pbx3 outkey=pbx1 context=global_pbx_transfer host=pbx3.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.234 and this on box2: [pbx1] type=friend auth=rsa inkeys=pbx1 outkey=pbx2 context=global_pbx_transfer host=pbx1.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.203 [pbx2] type=friend auth=rsa inkeys=pbx2 outkey=pbx2 context=global_pbx_transfer host=pbx2.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.233 [pbx3] type=friend auth=rsa inkeys=pbx3 outkey=pbx2 context=global_pbx_transfer host=pbx3.ipt.yyy.com deny=0.0.0.0 permit=xxx.187.142.234 And for some reason the calling system is sending pbx3 as the username.... Why would it do that? -----Original Message----- From: Steve Totaro [mailto:stotaro@asteriskhelpdesk.com] Sent: Saturday, March 25, 2006 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing No, you need to use different names. You can use friend rather than having separate entries for in/out. What do you get when you type iax2 show peers? You should be able to use friend and the same three entries on each box with the exception of changing the IP addresses. _____ From: Douglas Garstang [mailto:dgarstang@oneeighty.com] Sent: Saturday, March 25, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RE: IAX Incoming/Outgoing 12 hours later... still playing with this. Anyone got any ideas? Doug. -----Original Message----- From: Douglas Garstang Sent: Friday, March 24, 2006 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: IAX Incoming/Outgoing I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make sense any longer. Has anyone got a working example they could supply? Can I do all this with just three peers and one username? Thanks... Doug. [pbx1_inbound] type=user auth=rsa inkeys=pbx1 username=pbx1_inbound deny=0.0.0.0 permit=xxx.187.142.203 context=global_pbx_transfer [pbx1_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx1.ipt.yyy.com [pbx2_inbound] type=user auth=rsa inkeys=pbx2 username=pbx2_inbound deny=0.0.0.0 permit=xxx.187.142.204 context=global_pbx_transfer [pbx2_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx2.ipt.yyy.com [pbx3_inbound] type=user auth=rsa inkeys=pbx3 username=pbx3_inbound deny=0.0.0.0 permit=xxx.187.142.234 context=global_pbx_transfer [pbx3_outbound] type=peer auth=rsa outkey=pbx1 username=pbx3 host=pbx3.ipt.yyy.com -----Original Message----- From: George Vagenas [mailto:gvagasterisk@gmail.com] Sent: Fri 3/24/2006 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: George Vagenas Subject: Re: [Asterisk-Users] SIP trunk problem Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. On 3/25/06, Martin Joseph < ast@stillnewt.org> wrote: On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:> Hi all, > > I have the following problem, working with a SIP provider, if i setup > my SJPhone to register directly to their STUN server and working over > a 384/128 ADSL i have a really good quality, but then if i configure > Asterisk to register to the same provider over the same 384/128 > circuit the quality is REALLY BAD. The obvious difference is that > using directly the SJPhone i am using STUN, while when i am using > Asterisk to connect to my SIP provider and the SJPhone to connect to > Asterisk i have the following configuration for Asterisk. > > > register => user:pass@sip.provider.com > > [mysip] > host= sip.provider.com > type=peer > qualify=yes > username=user > secret=pass > nat=yes > disallow=all > allow=ulaw > > > I am using Asterisk 1.2.3. > > I think that i am missing something or misconfigure something because > for sure its not matter of the ADSL since in both tests i am doing i > am using the same circuit. > > Any idea please????I don't think using ulaw on a 128K bit upstream circuit is a good choice. I would use g729. Marty PS I can't be the stun server if asterisk is working, but quality is poor. _______________________________________________ --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> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060325/f7e3e1cb/attachment.htm
This is INSANE! My calling system has this iax.conf:
[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx1
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203
[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx1
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233
[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx1
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234
and here's how I am dialling PBX2... as you can see I am dialling _PBX2_:
exten => s-CHANUNAVAIL,1,Dial( IAX2/pbx2/${ARG1}@global_pbx_transfer,25,g)
When I run an iax debug on the caller, I see
VERSION : 2
CALLED NUMBER : 2944099
CODEC_PREFS : (ulaw|g729)
CALLING NUMBER : 2944093
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : Foo
LANGUAGE : en
CALLED CONTEXT : global_pbx_transfer
FORMAT : 4
CAPABILITY : 65535
ADSICPE : 2
DATE TIME : 2006-03-25 11:24:58
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00005ms SCall: 00004 DCall: 00006 [216.187.142.204:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ
Timestamp: 00010ms SCall: 00004 DCall: 00006 [216.187.142.204:4569]
AUTHMETHODS : 4
CHALLENGE : 627190238
USERNAME : pbx3
What on gods green earth would possibly make asterisk want to send a username of
PBX3???
-----Original Message-----
From: Douglas Garstang
Sent: Saturday, March 25, 2006 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing
Well, right now I have this on box1:
[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx1
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203
[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx1
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233
[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx1
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234
and this on box2:
[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx2
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203
[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx2
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233
[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx2
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234
And for some reason the calling system is sending pbx3 as the username.... Why
would it do that?
-----Original Message-----
From: Steve Totaro [mailto:stotaro@asteriskhelpdesk.com]
Sent: Saturday, March 25, 2006 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing
No, you need to use different names. You can use friend rather than having
separate entries for in/out. What do you get when you type iax2 show peers?
You should be able to use friend and the same three entries on each box with the
exception of changing the IP addresses.
_____
From: Douglas Garstang [mailto:dgarstang@oneeighty.com]
Sent: Saturday, March 25, 2006 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] RE: IAX Incoming/Outgoing
12 hours later... still playing with this. Anyone got any ideas?
Doug.
-----Original Message-----
From: Douglas Garstang
Sent: Friday, March 24, 2006 10:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users
Mailing List - Non-Commercial Discussion
Subject: IAX Incoming/Outgoing
I'vce got three Asterisk systems here that I'd like to be able to place
calls between with IAX. As usual, I've spent several hours playing with it,
really getting nowhere. Asterisk is so mentally draining. Each system, pbx1,
pbx2, pbx3, should be able to connect to every other. Do I need separate
user/peers or can I combine them into a single user=friend for each system? if I
place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say
that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make
sense any longer.
Has anyone got a working example they could supply? Can I do all this with just
three peers and one username?
Thanks... Doug.
[pbx1_inbound]
type=user
auth=rsa
inkeys=pbx1
username=pbx1_inbound
deny=0.0.0.0
permit=xxx.187.142.203
context=global_pbx_transfer
[pbx1_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx1.ipt.yyy.com
[pbx2_inbound]
type=user
auth=rsa
inkeys=pbx2
username=pbx2_inbound
deny=0.0.0.0
permit=xxx.187.142.204
context=global_pbx_transfer
[pbx2_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx2.ipt.yyy.com
[pbx3_inbound]
type=user
auth=rsa
inkeys=pbx3
username=pbx3_inbound
deny=0.0.0.0
permit=xxx.187.142.234
context=global_pbx_transfer
[pbx3_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx3
host=pbx3.ipt.yyy.com
-----Original Message-----
From: George Vagenas [mailto:gvagasterisk@gmail.com]
Sent: Fri 3/24/2006 10:30 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: George Vagenas
Subject: Re: [Asterisk-Users] SIP trunk problem
Marty,
But with the same 128 bit upstream circuit, directly connecting the SJPhone the
Stun server and using ulaw, everything is perfect. The problem comes when i am
putting Asterisk in the picture.
On 3/25/06, Martin Joseph < ast@stillnewt.org> wrote:
On Mar 24, 2006, at 1:19 PM, George Vagenas wrote:
> Hi all,
>
> I have the following problem, working with a SIP provider, if i setup
> my SJPhone to register directly to their STUN server and working over
> a 384/128 ADSL i have a really good quality, but then if i configure
> Asterisk to register to the same provider over the same 384/128
> circuit the quality is REALLY BAD. The obvious difference is that
> using directly the SJPhone i am using STUN, while when i am using
> Asterisk to connect to my SIP provider and the SJPhone to connect to
> Asterisk i have the following configuration for Asterisk.
>
>
> register => user:pass@sip.provider.com
>
> [mysip]
> host= sip.provider.com
> type=peer
> qualify=yes
> username=user
> secret=pass
> nat=yes
> disallow=all
> allow=ulaw
>
>
> I am using Asterisk 1.2.3.
>
> I think that i am missing something or misconfigure something because
> for sure its not matter of the ADSL since in both tests i am doing i
> am using the same circuit.
>
> Any idea please????
I don't think using ulaw on a 128K bit upstream circuit is a good
choice. I would use g729.
Marty
PS I can't be the stun server if asterisk is working, but quality is
poor.
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Well, I just tried your approach. I broke them all up into users/peers. Now it
makes even LESS sense. The pbx1 system is connecting to the pbx2 system, and
according to the iax debug, is sending a username of 'pbx3_in'. *lol*
[pbx1_in]
type=user
auth=rsa
inkeys=pbx1
context=global_pbx_transfer
deny=0.0.0.0
permit=xxx.187.142.203
[pbx1_out]
type=peer
auth=rsa
outkey=pbx1
host=pbx1.ipt.yyy.com
[pbx2_in]
type=user
auth=rsa
inkeys=pbx2
context=global_pbx_transfer
deny=0.0.0.0
permit=xxx.187.142.204
[pbx2_out]
type=peer
auth=rsa
outkey=pbx1
host=pbx2.ipt.yyy.com
[pbx3_in]
type=user
auth=rsa
inkeys=pbx3
context=global_pbx_transfer
deny=0.0.0.0
permit=xxx.187.142.234
[pbx3_out]
type=peer
auth=rsa
outkey=pbx1
host=pbx3.ipt.yyy.com
Here's how I connect:
exten => s-CHANUNAVAIL,1,Dial(IAX2/pbx2_out/${ARG1}@global_pbx_transfer,25,g)
and here's the IAX debug:
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW
Timestamp: 00003ms SCall: 00001 DCall: 00000 [xxx.187.142.204:4569]
VERSION : 2
CALLED NUMBER : 2944099
CODEC_PREFS : (ulaw|g729)
CALLING NUMBER : 2944093
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : Foo
LANGUAGE : en
CALLED CONTEXT : global_pbx_transfer
FORMAT : 4
CAPABILITY : 65535
ADSICPE : 2
DATE TIME : 2006-03-25 11:54:36
hestia*CLI>
-- Called pbx2_out/2944099@global_pbx_transfer
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00003ms SCall: 00002 DCall: 00001 [xxx.187.142.204:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ
Timestamp: 00005ms SCall: 00002 DCall: 00001 [xxx.187.142.204:4569]
AUTHMETHODS : 4
CHALLENGE : 129428696
USERNAME : pbx3_in <---- WHAT THE HELL IS THIS DOING
HERE?
> -----Original Message-----
> From: Brian Capouch [mailto:brianc@palaver.net]
> Sent: Saturday, March 25, 2006 11:46 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing
>
>
> Douglas Garstang wrote:
> > This is INSANE! My calling system has this iax.conf:
> >
>
> Search the archives for mails about separating
> originations/terminations
> by removing all friends and setting up the various
> interoperating boxes
> in a peer-user arrangement.
>
> I am pretty certain there are archived mails that urge people who use
> IAX to do that, and indicating that the various possible ambiguities
> with IAX friends is not a Good Thing.
>
> That would seem borne out by your experiences.
>
> I would also follow the time-honored programming technique of
> removing
> many of your constraints (keys, allow/disallows, etc.) in order to
> remove as many causes of uncertainty as possible. Then once
> the boxes
> are talking those things can be added back in a controlled
> manner. To
> my eyes your configurations have an awful lot of variable factors.
>
> Just where the insanity lies is another issue, which I don't
> care to get
> into at the present time :-)
>
> B.
> _______________________________________________
> --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
>
Why do I need a username at all if I am doing rsa authentication? Why doesn't it match against the key?> -----Original Message----- > From: Joshua Colp [mailto:joshnet@nbnet.nb.ca] > Sent: Saturday, March 25, 2006 12:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing > > > You do realize you're not sending along a username so it's > using another method to try to discover the username you're > trying to authenticate as on the server side? Apparently not. > > IAX2/username_to_use@peer_entry_to_use/extension@context > > Joshua Colp > > ----- Original Message ----- > From: Douglas Garstang > [mailto:dgarstang@oneeighty.com] > To: Asterisk Users Mailing List - > Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] > Sent: > Sat, 25 Mar 2006 14:55:28 -0400 > Subject: RE: [Asterisk-Users] RE: IAX > Incoming/Outgoing > > > > Well, I just tried your approach. I broke them all up into > users/peers. Now > > it makes even LESS sense. The pbx1 system is connecting to > the pbx2 system, > > and according to the iax debug, is sending a username of > 'pbx3_in'. *lol* > > > > [pbx1_in] > > type=user > > auth=rsa > > inkeys=pbx1 > > context=global_pbx_transfer > > deny=0.0.0.0 > > permit=xxx.187.142.203 > > > > [pbx1_out] > > type=peer > > auth=rsa > > outkey=pbx1 > > host=pbx1.ipt.yyy.com > > > > [pbx2_in] > > type=user > > auth=rsa > > inkeys=pbx2 > > context=global_pbx_transfer > > deny=0.0.0.0 > > permit=xxx.187.142.204 > > > > [pbx2_out] > > type=peer > > auth=rsa > > outkey=pbx1 > > host=pbx2.ipt.yyy.com > > > > [pbx3_in] > > type=user > > auth=rsa > > inkeys=pbx3 > > context=global_pbx_transfer > > deny=0.0.0.0 > > permit=xxx.187.142.234 > > > > [pbx3_out] > > type=peer > > auth=rsa > > outkey=pbx1 > > host=pbx3.ipt.yyy.com > > > > Here's how I connect: > > exten => > > s-CHANUNAVAIL,1,Dial(IAX2/pbx2_out/${ARG1}@global_pbx_transfer,25,g) > > > > and here's the IAX debug: > > Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX > Subclass: NEW > > > > Timestamp: 00003ms SCall: 00001 DCall: 00000 > [xxx.187.142.204:4569] > > VERSION : 2 > > CALLED NUMBER : 2944099 > > CODEC_PREFS : (ulaw|g729) > > CALLING NUMBER : 2944093 > > CALLING PRESNTN : 0 > > CALLING TYPEOFN : 0 > > CALLING TRANSIT : 0 > > CALLING NAME : Foo > > LANGUAGE : en > > CALLED CONTEXT : global_pbx_transfer > > FORMAT : 4 > > CAPABILITY : 65535 > > ADSICPE : 2 > > DATE TIME : 2006-03-25 11:54:36 > > hestia*CLI> > > -- Called pbx2_out/2944099@global_pbx_transfer > > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX > Subclass: ACK > > > > Timestamp: 00003ms SCall: 00002 DCall: 00001 > [xxx.187.142.204:4569] > > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX > Subclass: > > AUTHREQ > > Timestamp: 00005ms SCall: 00002 DCall: 00001 > [xxx.187.142.204:4569] > > AUTHMETHODS : 4 > > CHALLENGE : 129428696 > > USERNAME : pbx3_in <---- WHAT THE HELL > IS THIS DOING > > HERE? > > > > > > > > > > > -----Original Message----- > > > From: Brian Capouch [mailto:brianc@palaver.net] > > > Sent: Saturday, March 25, 2006 11:46 AM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing > > > > > > > > > Douglas Garstang wrote: > > > > This is INSANE! My calling system has this iax.conf: > > > > > > > > > > Search the archives for mails about separating > > > originations/terminations > > > by removing all friends and setting up the various > > > interoperating boxes > > > in a peer-user arrangement. > > > > > > I am pretty certain there are archived mails that urge > people who use > > > IAX to do that, and indicating that the various possible > ambiguities > > > with IAX friends is not a Good Thing. > > > > > > That would seem borne out by your experiences. > > > > > > I would also follow the time-honored programming technique of > > > removing > > > many of your constraints (keys, allow/disallows, etc.) in > order to > > > remove as many causes of uncertainty as possible. Then once > > > the boxes > > > are talking those things can be added back in a controlled > > > manner. To > > > my eyes your configurations have an awful lot of variable factors. > > > > > > Just where the insanity lies is another issue, which I don't > > > care to get > > > into at the present time :-) > > > > > > B. > > > _______________________________________________ > > > --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 > > > _______________________________________________ > --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, actually I did... sort of... I picked a random post and hit reply as I normally do. I forgot to clear the old text, but it's obvious that in no way it detracted from the other person's post, and the use of the word 'hijack' is highly dubious.. Doug. -----Original Message----- From: Martin Joseph [mailto:ast@stillnewt.org] Sent: Saturday, March 25, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing On Mar 25, 2006, at 2:01 PM, Steve Totaro wrote: Just a few things Doug and they are just constructive criticism so don?t take them the wrong way. 1 You hijacked some else?s thread about a SIP trunk problem. Very frowned upon and will decrease people willing to help.. This does not appear to be true to me. Perhaps it's your crappy mail reader. Marty -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060325/fb7a5366/attachment.htm
I could ask why it can't authenticate against the key, but we've already been there. So, if I have 5 asterisk systems, and I want to have a different key on each, and each system has a user and a peer section, and I have to use different usernames... oh boy... this sounds like a horrible mess.> -----Original Message----- > From: Joshua Colp [mailto:joshnet@nbnet.nb.ca] > Sent: Saturday, March 25, 2006 12:19 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing > > > It still needs to know the username so it knows what entry in > iax.conf to use for that information, such as the key to use. > > Joshua Colp > > ----- Original Message ----- > From: Douglas Garstang > [mailto:dgarstang@oneeighty.com] > To: Asterisk Users Mailing List - > Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] > Sent: > Sat, 25 Mar 2006 15:15:24 -0400 > Subject: RE: [Asterisk-Users] RE: IAX > Incoming/Outgoing > > > > Why do I need a username at all if I am doing rsa > authentication? Why > > doesn't it match against the key? > > > > > -----Original Message----- > > > From: Joshua Colp [mailto:joshnet@nbnet.nb.ca] > > > Sent: Saturday, March 25, 2006 12:11 PM > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing > > > > > > > > > You do realize you're not sending along a username so it's > > > using another method to try to discover the username you're > > > trying to authenticate as on the server side? Apparently not. > > > > > > IAX2/username_to_use@peer_entry_to_use/extension@context > > > > > > Joshua Colp > > > > > > ----- Original Message ----- > > > From: Douglas Garstang > > > [mailto:dgarstang@oneeighty.com] > > > To: Asterisk Users Mailing List - > > > Non-Commercial Discussion [mailto:asterisk-users@lists.digium.com] > > > Sent: > > > Sat, 25 Mar 2006 14:55:28 -0400 > > > Subject: RE: [Asterisk-Users] RE: IAX > > > Incoming/Outgoing > > > > > > > > > > Well, I just tried your approach. I broke them all up into > > > users/peers. Now > > > > it makes even LESS sense. The pbx1 system is connecting to > > > the pbx2 system, > > > > and according to the iax debug, is sending a username of > > > 'pbx3_in'. *lol* > > > > > > > > [pbx1_in] > > > > type=user > > > > auth=rsa > > > > inkeys=pbx1 > > > > context=global_pbx_transfer > > > > deny=0.0.0.0 > > > > permit=xxx.187.142.203 > > > > > > > > [pbx1_out] > > > > type=peer > > > > auth=rsa > > > > outkey=pbx1 > > > > host=pbx1.ipt.yyy.com > > > > > > > > [pbx2_in] > > > > type=user > > > > auth=rsa > > > > inkeys=pbx2 > > > > context=global_pbx_transfer > > > > deny=0.0.0.0 > > > > permit=xxx.187.142.204 > > > > > > > > [pbx2_out] > > > > type=peer > > > > auth=rsa > > > > outkey=pbx1 > > > > host=pbx2.ipt.yyy.com > > > > > > > > [pbx3_in] > > > > type=user > > > > auth=rsa > > > > inkeys=pbx3 > > > > context=global_pbx_transfer > > > > deny=0.0.0.0 > > > > permit=xxx.187.142.234 > > > > > > > > [pbx3_out] > > > > type=peer > > > > auth=rsa > > > > outkey=pbx1 > > > > host=pbx3.ipt.yyy.com > > > > > > > > Here's how I connect: > > > > exten => > > > > > s-CHANUNAVAIL,1,Dial(IAX2/pbx2_out/${ARG1}@global_pbx_transfer,25,g) > > > > > > > > and here's the IAX debug: > > > > Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX > > > Subclass: NEW > > > > > > > > Timestamp: 00003ms SCall: 00001 DCall: 00000 > > > [xxx.187.142.204:4569] > > > > VERSION : 2 > > > > CALLED NUMBER : 2944099 > > > > CODEC_PREFS : (ulaw|g729) > > > > CALLING NUMBER : 2944093 > > > > CALLING PRESNTN : 0 > > > > CALLING TYPEOFN : 0 > > > > CALLING TRANSIT : 0 > > > > CALLING NAME : Foo > > > > LANGUAGE : en > > > > CALLED CONTEXT : global_pbx_transfer > > > > FORMAT : 4 > > > > CAPABILITY : 65535 > > > > ADSICPE : 2 > > > > DATE TIME : 2006-03-25 11:54:36 > > > > hestia*CLI> > > > > -- Called pbx2_out/2944099@global_pbx_transfer > > > > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX > > > Subclass: ACK > > > > > > > > Timestamp: 00003ms SCall: 00002 DCall: 00001 > > > [xxx.187.142.204:4569] > > > > Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX > > > Subclass: > > > > AUTHREQ > > > > Timestamp: 00005ms SCall: 00002 DCall: 00001 > > > [xxx.187.142.204:4569] > > > > AUTHMETHODS : 4 > > > > CHALLENGE : 129428696 > > > > USERNAME : pbx3_in <---- WHAT THE HELL > > > IS THIS DOING > > > > HERE? > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > > > From: Brian Capouch [mailto:brianc@palaver.net] > > > > > Sent: Saturday, March 25, 2006 11:46 AM > > > > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > > Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing > > > > > > > > > > > > > > > Douglas Garstang wrote: > > > > > > This is INSANE! My calling system has this iax.conf: > > > > > > > > > > > > > > > > Search the archives for mails about separating > > > > > originations/terminations > > > > > by removing all friends and setting up the various > > > > > interoperating boxes > > > > > in a peer-user arrangement. > > > > > > > > > > I am pretty certain there are archived mails that urge > > > people who use > > > > > IAX to do that, and indicating that the various possible > > > ambiguities > > > > > with IAX friends is not a Good Thing. > > > > > > > > > > That would seem borne out by your experiences. > > > > > > > > > > I would also follow the time-honored programming technique of > > > > > removing > > > > > many of your constraints (keys, allow/disallows, etc.) in > > > order to > > > > > remove as many causes of uncertainty as possible. Then once > > > > > the boxes > > > > > are talking those things can be added back in a controlled > > > > > manner. To > > > > > my eyes your configurations have an awful lot of > variable factors. > > > > > > > > > > Just where the insanity lies is another issue, which I don't > > > > > care to get > > > > > into at the present time :-) > > > > > > > > > > B. > > > > > _______________________________________________ > > > > > --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 > > > > > > > _______________________________________________ > > > --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 > > > _______________________________________________ > --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 >
No, you need to use different names. You can use friend rather than having separate entries for in/out. What do you get when you type iax2 show peers? You should be able to use friend and the same three entries on each box with the exception of changing the IP addresses. _____ From: Douglas Garstang [mailto:dgarstang@oneeighty.com] Sent: Saturday, March 25, 2006 12:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] RE: IAX Incoming/Outgoing 12 hours later... still playing with this. Anyone got any ideas? Doug. -----Original Message----- From: Douglas Garstang Sent: Friday, March 24, 2006 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: IAX Incoming/Outgoing I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a call from pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2 will look for a [pbx1_outbound] .... oh dear... this doesn't make sense any longer. Has anyone got a working example they could supply? Can I do all this with just three peers and one username? Thanks... Doug. [pbx1_inbound] type=user auth=rsa inkeys=pbx1 username=pbx1_inbound deny=0.0.0.0 permit=xxx.187.142.203 context=global_pbx_transfer [pbx1_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx1.ipt.yyy.com [pbx2_inbound] type=user auth=rsa inkeys=pbx2 username=pbx2_inbound deny=0.0.0.0 permit=xxx.187.142.204 context=global_pbx_transfer [pbx2_outbound] type=peer auth=rsa outkey=pbx1 username=pbx1 host=pbx2.ipt.yyy.com [pbx3_inbound] type=user auth=rsa inkeys=pbx3 username=pbx3_inbound deny=0.0.0.0 permit=xxx.187.142.234 context=global_pbx_transfer [pbx3_outbound] type=peer auth=rsa outkey=pbx1 username=pbx3 host=pbx3.ipt.yyy.com -----Original Message----- From: George Vagenas [mailto:gvagasterisk@gmail.com] Sent: Fri 3/24/2006 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: George Vagenas Subject: Re: [Asterisk-Users] SIP trunk problem Marty, But with the same 128 bit upstream circuit, directly connecting the SJPhone the Stun server and using ulaw, everything is perfect. The problem comes when i am putting Asterisk in the picture. On 3/25/06, Martin Joseph <ast@stillnewt.org> wrote: On Mar 24, 2006, at 1:19 PM, George Vagenas wrote: > Hi all, > > I have the following problem, working with a SIP provider, if i setup > my SJPhone to register directly to their STUN server and working over > a 384/128 ADSL i have a really good quality, but then if i configure > Asterisk to register to the same provider over the same 384/128 > circuit the quality is REALLY BAD. The obvious difference is that > using directly the SJPhone i am using STUN, while when i am using > Asterisk to connect to my SIP provider and the SJPhone to connect to > Asterisk i have the following configuration for Asterisk. > > > register => user:pass@sip.provider.com > > [mysip] > host=sip.provider.com > type=peer > qualify=yes > username=user > secret=pass > nat=yes > disallow=all > allow=ulaw > > > I am using Asterisk 1.2.3. > > I think that i am missing something or misconfigure something because > for sure its not matter of the ADSL since in both tests i am doing i > am using the same circuit. > > Any idea please???? I don't think using ulaw on a 128K bit upstream circuit is a good choice. I would use g729. Marty PS I can't be the stun server if asterisk is working, but quality is poor. _______________________________________________ --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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060325/7a4a4064/attachment.htm
Just a few things Doug and they are just constructive criticism so don't
take them the wrong way.
1. You hijacked some else's thread about a SIP trunk problem. Very
frowned upon and will decrease people willing to help..
2. All of your posts are so dramatic and many times negative which
will also decrease willing help.
3. You are posting way too much without experimenting and thinking
things through. Take the list as a place to post knowledge and a place
to get answers when you have tried everything you can think of.
I had a rule to put your emails directly in my deleted items folder from
the first day you started posting to this list totally bashing asterisk
and the community. I recently had to re-do my machine so the rule was
lost. I am hoping that I don't have to put it back in place.
Now back to your problem.
Simplify your conf. Remove the keys and use secret
Change your dial statement to Dial(iax2/username:secret@ipaddress)
Are you dialing from PBX3?
Thanks,
Steve Totaro
_____
From: Douglas Garstang [mailto:dgarstang@oneeighty.com]
Sent: Saturday, March 25, 2006 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing
This is INSANE! My calling system has this iax.conf:
[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx1
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203
[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx1
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233
[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx1
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234
and here's how I am dialling PBX2... as you can see I am dialling
_PBX2_:
exten => s-CHANUNAVAIL,1,Dial(IAX2/pbx2/${ARG1}@global_pbx_transfer,25,g
<mailto:IAX2/pbx2/$%7bARG1%7d@global_pbx_transfer,25,g> )
When I run an iax debug on the caller, I see
VERSION : 2
CALLED NUMBER : 2944099
CODEC_PREFS : (ulaw|g729)
CALLING NUMBER : 2944093
CALLING PRESNTN : 0
CALLING TYPEOFN : 0
CALLING TRANSIT : 0
CALLING NAME : Foo
LANGUAGE : en
CALLED CONTEXT : global_pbx_transfer
FORMAT : 4
CAPABILITY : 65535
ADSICPE : 2
DATE TIME : 2006-03-25 11:24:58
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
ACK
Timestamp: 00005ms SCall: 00004 DCall: 00006 [216.187.142.204:4569]
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
AUTHREQ
Timestamp: 00010ms SCall: 00004 DCall: 00006 [216.187.142.204:4569]
AUTHMETHODS : 4
CHALLENGE : 627190238
USERNAME : pbx3
What on gods green earth would possibly make asterisk want to send a
username of PBX3???
-----Original Message-----
From: Douglas Garstang
Sent: Saturday, March 25, 2006 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing
Well, right now I have this on box1:
[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx1
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203
[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx1
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233
[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx1
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234
and this on box2:
[pbx1]
type=friend
auth=rsa
inkeys=pbx1
outkey=pbx2
context=global_pbx_transfer
host=pbx1.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.203
[pbx2]
type=friend
auth=rsa
inkeys=pbx2
outkey=pbx2
context=global_pbx_transfer
host=pbx2.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.233
[pbx3]
type=friend
auth=rsa
inkeys=pbx3
outkey=pbx2
context=global_pbx_transfer
host=pbx3.ipt.yyy.com
deny=0.0.0.0
permit=xxx.187.142.234
And for some reason the calling system is sending pbx3 as the
username.... Why would it do that?
-----Original Message-----
From: Steve Totaro [mailto:stotaro@asteriskhelpdesk.com]
Sent: Saturday, March 25, 2006 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing
No, you need to use different names. You can use friend
rather than having separate entries for in/out. What do you get when
you type iax2 show peers? You should be able to use friend and the same
three entries on each box with the exception of changing the IP
addresses.
_____
From: Douglas Garstang [mailto:dgarstang@oneeighty.com]
Sent: Saturday, March 25, 2006 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: [Asterisk-Users] RE: IAX Incoming/Outgoing
12 hours later... still playing with this. Anyone got
any ideas?
Doug.
-----Original Message-----
From: Douglas Garstang
Sent: Friday, March 24, 2006 10:53 PM
To: Asterisk Users Mailing List - Non-Commercial
Discussion; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: IAX Incoming/Outgoing
I'vce got three Asterisk systems here that I'd
like to be able to place calls between with IAX. As usual, I've spent
several hours playing with it, really getting nowhere. Asterisk is so
mentally draining. Each system, pbx1, pbx2, pbx3, should be able to
connect to every other. Do I need separate user/peers or can I combine
them into a single user=friend for each system? if I place a call from
pbx1 to pbx2 as pbx1_outbound, it should work.... the docs say that pbx2
will look for a [pbx1_outbound] .... oh dear... this doesn't make sense
any longer.
Has anyone got a working example they could
supply? Can I do all this with just three peers and one username?
Thanks... Doug.
[pbx1_inbound]
type=user
auth=rsa
inkeys=pbx1
username=pbx1_inbound
deny=0.0.0.0
permit=xxx.187.142.203
context=global_pbx_transfer
[pbx1_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx1.ipt.yyy.com
[pbx2_inbound]
type=user
auth=rsa
inkeys=pbx2
username=pbx2_inbound
deny=0.0.0.0
permit=xxx.187.142.204
context=global_pbx_transfer
[pbx2_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx1
host=pbx2.ipt.yyy.com
[pbx3_inbound]
type=user
auth=rsa
inkeys=pbx3
username=pbx3_inbound
deny=0.0.0.0
permit=xxx.187.142.234
context=global_pbx_transfer
[pbx3_outbound]
type=peer
auth=rsa
outkey=pbx1
username=pbx3
host=pbx3.ipt.yyy.com
-----Original Message-----
From: George Vagenas
[mailto:gvagasterisk@gmail.com]
Sent: Fri 3/24/2006 10:30 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Cc: George Vagenas
Subject: Re: [Asterisk-Users] SIP trunk
problem
Marty,
But with the same 128 bit upstream
circuit, directly connecting the SJPhone the Stun server and using ulaw,
everything is perfect. The problem comes when i am putting Asterisk in
the picture.
On 3/25/06, Martin Joseph
<ast@stillnewt.org> wrote:
On Mar 24, 2006, at 1:19 PM, George
Vagenas wrote:
> Hi all,
>
> I have the following problem, working
with a SIP provider, if i setup
> my SJPhone to register directly to
their STUN server and working over
> a 384/128 ADSL i have a really good
quality, but then if i configure
> Asterisk to register to the same
provider over the same 384/128
> circuit the quality is REALLY BAD. The
obvious difference is that
> using directly the SJPhone i am using
STUN, while when i am using
> Asterisk to connect to my SIP provider
and the SJPhone to connect to
> Asterisk i have the following
configuration for Asterisk.
>
>
> register =>
user:pass@sip.provider.com
>
> [mysip]
> host=sip.provider.com
> type=peer
> qualify=yes
> username=user
> secret=pass
> nat=yes
> disallow=all
> allow=ulaw
>
>
> I am using Asterisk 1.2.3.
>
> I think that i am missing something
or misconfigure something because
> for sure its not matter of the ADSL
since in both tests i am doing i
> am using the same circuit.
>
> Any idea please????
I don't think using ulaw on a 128K bit
upstream circuit is a good
choice. I would use g729.
Marty
PS I can't be the stun server if
asterisk is working, but quality is
poor.
_______________________________________________
--Bandwidth and Colocation provided by
Easynews.com --
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Not to hijack but... http://en.wikipedia.org/wiki/Thread_Hijacking _____ From: Douglas Garstang [mailto:dgarstang@oneeighty.com] Sent: Saturday, March 25, 2006 2:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing Well, actually I did... sort of... I picked a random post and hit reply as I normally do. I forgot to clear the old text, but it's obvious that in no way it detracted from the other person's post, and the use of the word 'hijack' is highly dubious.. Doug. -----Original Message----- From: Martin Joseph [mailto:ast@stillnewt.org] Sent: Saturday, March 25, 2006 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing On Mar 25, 2006, at 2:01 PM, Steve Totaro wrote: Just a few things Doug and they are just constructive criticism so don't take them the wrong way. 1 You hijacked some else's thread about a SIP trunk problem. Very frowned upon and will decrease people willing to help.. This does not appear to be true to me. Perhaps it's your crappy mail reader. Marty -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060325/f29546d8/attachment.htm
Maybe you are better off with dundi ? -- Michiel van Baak michiel@vanbaak.info http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?"
> I could ask why it can't authenticate against the key, but we've already been > there. > > So, if I have 5 asterisk systems, and I want to have a different key on each, > and each system has a user and a peer section, and I have to use different > usernames... oh boy... this sounds like a horrible mess.I've been using a setup of one user for "incoming" and many outgoing peers. I'm not sure what the other poster meant that you can't do this. It works just fine. One thing I'll mention, and maybe if the developers are reading they can comment if this has changed, but in 1.0.x, and versions of CVS up to at least 05/2005, changes to the users and peers in iax.conf would often require a full restart to take effect. I don't use RSA since my IAX links all go over IPSec tunnels, but here's what my users and peers look like: [iax-in] type=user secret=XXXX context=extensions trunk=no tos=0x04 disallow=all allow=gsm [ast551-out] type=peer secret=XXXX username=ast551 host=XX.XX.XX.XX qualify=1000 disallow=all allow=gsm trunk=no tos=0x04 [ast129-out] type=peer secret=XXXX username=ast129 host=YY.YY.YY.YY qualify=1000 disallow=all allow=gsm trunk=no tos=0x04 etc.... - Noah>> -----Original Message----- >> From: Joshua Colp [mailto:joshnet at nbnet.nb.ca] >> Sent: Saturday, March 25, 2006 12:19 PM >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing >> >> >> It still needs to know the username so it knows what entry in >> iax.conf to use for that information, such as the key to use. >> >> Joshua Colp >> >> ----- Original Message ----- >> From: Douglas Garstang >> [mailto:dgarstang at oneeighty.com] >> To: Asterisk Users Mailing List - >> Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com] >> Sent: >> Sat, 25 Mar 2006 15:15:24 -0400 >> Subject: RE: [Asterisk-Users] RE: IAX >> Incoming/Outgoing >> >> >>> Why do I need a username at all if I am doing rsa >> authentication? Why >>> doesn't it match against the key? >>> >>>> -----Original Message----- >>>> From: Joshua Colp [mailto:joshnet at nbnet.nb.ca] >>>> Sent: Saturday, March 25, 2006 12:11 PM >>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>> Subject: RE: [Asterisk-Users] RE: IAX Incoming/Outgoing >>>> >>>> >>>> You do realize you're not sending along a username so it's >>>> using another method to try to discover the username you're >>>> trying to authenticate as on the server side? Apparently not. >>>> >>>> IAX2/username_to_use at peer_entry_to_use/extension at context >>>> >>>> Joshua Colp >>>> >>>> ----- Original Message ----- >>>> From: Douglas Garstang >>>> [mailto:dgarstang at oneeighty.com] >>>> To: Asterisk Users Mailing List - >>>> Non-Commercial Discussion [mailto:asterisk-users at lists.digium.com] >>>> Sent: >>>> Sat, 25 Mar 2006 14:55:28 -0400 >>>> Subject: RE: [Asterisk-Users] RE: IAX >>>> Incoming/Outgoing >>>> >>>> >>>>> Well, I just tried your approach. I broke them all up into >>>> users/peers. Now >>>>> it makes even LESS sense. The pbx1 system is connecting to >>>> the pbx2 system, >>>>> and according to the iax debug, is sending a username of >>>> 'pbx3_in'. *lol* >>>>> >>>>> [pbx1_in] >>>>> type=user >>>>> auth=rsa >>>>> inkeys=pbx1 >>>>> context=global_pbx_transfer >>>>> deny=0.0.0.0 >>>>> permit=xxx.187.142.203 >>>>> >>>>> [pbx1_out] >>>>> type=peer >>>>> auth=rsa >>>>> outkey=pbx1 >>>>> host=pbx1.ipt.yyy.com >>>>> >>>>> [pbx2_in] >>>>> type=user >>>>> auth=rsa >>>>> inkeys=pbx2 >>>>> context=global_pbx_transfer >>>>> deny=0.0.0.0 >>>>> permit=xxx.187.142.204 >>>>> >>>>> [pbx2_out] >>>>> type=peer >>>>> auth=rsa >>>>> outkey=pbx1 >>>>> host=pbx2.ipt.yyy.com >>>>> >>>>> [pbx3_in] >>>>> type=user >>>>> auth=rsa >>>>> inkeys=pbx3 >>>>> context=global_pbx_transfer >>>>> deny=0.0.0.0 >>>>> permit=xxx.187.142.234 >>>>> >>>>> [pbx3_out] >>>>> type=peer >>>>> auth=rsa >>>>> outkey=pbx1 >>>>> host=pbx3.ipt.yyy.com >>>>> >>>>> Here's how I connect: >>>>> exten => >>>>> >> s-CHANUNAVAIL,1,Dial(IAX2/pbx2_out/${ARG1}@global_pbx_transfer,25,g) >>>>> >>>>> and here's the IAX debug: >>>>> Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX >>>> Subclass: NEW >>>>> >>>>> Timestamp: 00003ms SCall: 00001 DCall: 00000 >>>> [xxx.187.142.204:4569] >>>>> VERSION : 2 >>>>> CALLED NUMBER : 2944099 >>>>> CODEC_PREFS : (ulaw|g729) >>>>> CALLING NUMBER : 2944093 >>>>> CALLING PRESNTN : 0 >>>>> CALLING TYPEOFN : 0 >>>>> CALLING TRANSIT : 0 >>>>> CALLING NAME : Foo >>>>> LANGUAGE : en >>>>> CALLED CONTEXT : global_pbx_transfer >>>>> FORMAT : 4 >>>>> CAPABILITY : 65535 >>>>> ADSICPE : 2 >>>>> DATE TIME : 2006-03-25 11:54:36 >>>>> hestia*CLI> >>>>> -- Called pbx2_out/2944099 at global_pbx_transfer >>>>> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX >>>> Subclass: ACK >>>>> >>>>> Timestamp: 00003ms SCall: 00002 DCall: 00001 >>>> [xxx.187.142.204:4569] >>>>> Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX >>>> Subclass: >>>>> AUTHREQ >>>>> Timestamp: 00005ms SCall: 00002 DCall: 00001 >>>> [xxx.187.142.204:4569] >>>>> AUTHMETHODS : 4 >>>>> CHALLENGE : 129428696 >>>>> USERNAME : pbx3_in <---- WHAT THE HELL >>>> IS THIS DOING >>>>> HERE? >>>>> >>>>> >>>>> >>>>> >>>>>> -----Original Message----- >>>>>> From: Brian Capouch [mailto:brianc at palaver.net] >>>>>> Sent: Saturday, March 25, 2006 11:46 AM >>>>>> To: Asterisk Users Mailing List - Non-Commercial Discussion >>>>>> Subject: Re: [Asterisk-Users] RE: IAX Incoming/Outgoing >>>>>> >>>>>> >>>>>> Douglas Garstang wrote: >>>>>>> This is INSANE! My calling system has this iax.conf: >>>>>>> >>>>>> >>>>>> Search the archives for mails about separating >>>>>> originations/terminations >>>>>> by removing all friends and setting up the various >>>>>> interoperating boxes >>>>>> in a peer-user arrangement. >>>>>> >>>>>> I am pretty certain there are archived mails that urge >>>> people who use >>>>>> IAX to do that, and indicating that the various possible >>>> ambiguities >>>>>> with IAX friends is not a Good Thing. >>>>>> >>>>>> That would seem borne out by your experiences. >>>>>> >>>>>> I would also follow the time-honored programming technique of >>>>>> removing >>>>>> many of your constraints (keys, allow/disallows, etc.) in >>>> order to >>>>>> remove as many causes of uncertainty as possible. Then once >>>>>> the boxes >>>>>> are talking those things can be added back in a controlled >>>>>> manner. To >>>>>> my eyes your configurations have an awful lot of >> variable factors. >>>>>> >>>>>> Just where the insanity lies is another issue, which I don't >>>>>> care to get >>>>>> into at the present time :-) >>>>>> >>>>>> B. >>>>>> _______________________________________________ >>>>>> --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 >>>>> >>>> _______________________________________________ >>>> --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 >>> >> _______________________________________________ >> --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