On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote:> > Thanks George. > > I will correct my local_net in the morning. > > Vitelity chan_sip settings I have working, do not have a fromuser. > sip.conf settings... > > I think you can actually specify anything, it just has to be populatedwith something other than a sub-account username.> [HVout] > > type=friend > > dtmfmode=auto > > host=64.2.142.93 > > disallow=all > > allow=ulaw > > canreinvite=no > > trustrpid=yes > > sendrpid=yes > > nat=yes > > context=TestApp > > > > On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph at fairview5.com> > wrote: > > > > On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: >> >> I am not sure if I entered the correct settings for the transport >> information. >> >> For the local_net, I entered my local ip address, but no mask. I will >> check with the network admin so he can verify the settings I entered. >> >> >> > You need the network and mask. For example if the ip address and mask of > the test machine is 192.168.0.1/255.255.255.0 then the correct entry > would be 192.168.0.0/24. > > >> One minor detail, we are using ip authentication. When Vitelity changed >> my account from user based authentication to IP based authentication, they >> stopped including a user for the account. >> >> >> >> Should these settings work without the from_user (IP based >> authentication) or do I need to get the account name from Vitelity? >> >> > You definitely need the master account login username. If you has this > working with chan_sip, then try the 'fromuser' from sip.conf and user is > from_user. > > > > >> >> >> Have a great day! >> >> >> >> Da >> >> >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *George Joseph >> *Sent:* Monday, December 15, 2014 7:27 PM >> *To:* Asterisk Users Mailing List - Non-Commercial Discussion >> *Subject:* Re: [asterisk-users] PJSIP configuration question >> >> >> >> Ok Dan, try this... I was able to get this to work behind a NAT and with >> ip address authentication. >> >> [global] >> type = global >> debug = yes >> >> [transport1] >> type = transport >> bind = 0.0.0.0 >> protocol = udp >> >> >> >> *local_net=<yourlocalnet I.E. 10.10.10.10/24 >> <http://10.10.10.10/24>>external_media_address=<your public ip >> address>external_signaling_address=<your public address>* >> [outbound.vitelity.net] >> type = aor >> remove_existing = yes >> qualify_frequency = 60 >> contact = sip:64.2.142.93 >> >> [outbound.vitelity.net] >> type = endpoint >> context = TestApp >> transport = transport1 >> aors = outbound.vitelity.net >> dtmf_mode = rfc4733 >> force_rport = yes >> rtp_symmetric = yes >> rewrite_contact = yes >> send_rpid = yes >> trust_id_inbound = yes >> disallow = all >> allow = ulaw >> direct_media = no >> >> *from_user=<your main vitelity account name> ; Not subaccount* >> >> [outbound.vitelity.net] >> type = identify >> endpoint = outbound.vitelity.net >> match = 64.2.142.93 >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> New to Asterisk? Join us for a live introductory webinar every Thurs: >> http://www.asterisk.org/hello >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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/20141215/992abdfd/attachment-0001.html>
Thanks George. I will give it a try. Have a great day! Dan From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of George Joseph Sent: Monday, December 15, 2014 11:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PJSIP configuration question I think you can actually specify anything, it just has to be populated with something other than a sub-account username. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141216/c2f4894b/attachment.html>
I corrected my local_net setting (based on advice from network admin).
I have tried several different values for the from_user and still have the same
problem.
Asterisk receives the OK from Vitelity.
Asterisk sends the ACK (without a Contact header).
Vitelity doesn?t seem to process it, so they send an OK again.
The OK receive, Transmit ACK occurs 4 times.
A short while later, Vitelity hangs up on my cell phone.
Asterisk is never told the call is gone.
If I hangup the call from Asterisk side,
Asterisk sends the BYE message.
Vitelity responds with a ?SIP/2.0 481 Call leg/transaction does not exist?
Again, the trace indicates the ACK message is missing the Contact header.
Additional note: the network admin is asking why the local_net,
external_media_address, and external_signalling_address are needed. He wrote
me??You should NOT have to know your public IP. The firewall should be doing
fixup commands to change the values in the packet?
From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces
at lists.digium.com] On Behalf Of George Joseph
Sent: Monday, December 15, 2014 11:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
Thanks George.
I will correct my local_net in the morning.
Vitelity chan_sip settings I have working, do not have a fromuser.
sip.conf settings...
I think you can actually specify anything, it just has to be populated with
something other than a sub-account username.
[HVout]
type=friend
dtmfmode=auto
host=64.2.142.93
disallow=all
allow=ulaw
canreinvite=no
trustrpid=yes
sendrpid=yes
nat=yes
context=TestApp
On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph at
fairview5.com<mailto:george.joseph at fairview5.com>> wrote:
On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com<mailto:dan
at amtelco.com>> wrote:
I am not sure if I entered the correct settings for the transport information.
For the local_net, I entered my local ip address, but no mask. I will check
with the network admin so he can verify the settings I entered.
You need the network and mask. For example if the ip address and mask of the
test machine is
192.168.0.1/255.255.255.0<http://192.168.0.1/255.255.255.0> then the
correct entry would be 192.168.0.0/24<http://192.168.0.0/24>.
One minor detail, we are using ip authentication. When Vitelity changed my
account from user based authentication to IP based authentication, they stopped
including a user for the account.
Should these settings work without the from_user (IP based authentication) or do
I need to get the account name from Vitelity?
You definitely need the master account login username. If you has this working
with chan_sip, then try the 'fromuser' from sip.conf and user is
from_user.
Have a great day!
Da
From: asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>
[mailto:asterisk-users-bounces at
lists.digium.com<mailto:asterisk-users-bounces at lists.digium.com>] On
Behalf Of George Joseph
Sent: Monday, December 15, 2014 7:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] PJSIP configuration question
Ok Dan, try this... I was able to get this to work behind a NAT and with ip
address authentication.
[global]
type = global
debug = yes
[transport1]
type = transport
bind = 0.0.0.0
protocol = udp
local_net=<yourlocalnet I.E. 10.10.10.10/24<http://10.10.10.10/24>>
external_media_address=<your public ip address>
external_signaling_address=<your public address>
[outbound.vitelity.net<http://outbound.vitelity.net>]
type = aor
remove_existing = yes
qualify_frequency = 60
contact = sip:64.2.142.93
[outbound.vitelity.net<http://outbound.vitelity.net>]
type = endpoint
context = TestApp
transport = transport1
aors = outbound.vitelity.net<http://outbound.vitelity.net>
dtmf_mode = rfc4733
force_rport = yes
rtp_symmetric = yes
rewrite_contact = yes
send_rpid = yes
trust_id_inbound = yes
disallow = all
allow = ulaw
direct_media = no
from_user=<your main vitelity account name> ; Not subaccount
[outbound.vitelity.net<http://outbound.vitelity.net>]
type = identify
endpoint = outbound.vitelity.net<http://outbound.vitelity.net>
match = 64.2.142.93
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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On Tue, Dec 16, 2014 at 9:00 AM, Dan Cropp <dan at amtelco.com> wrote:> > I corrected my local_net setting (based on advice from network admin). > > > > I have tried several different values for the from_user and still have the > same problem. > > > > Asterisk receives the OK from Vitelity. > > Asterisk sends the ACK (without a Contact header). > > Vitelity doesn?t seem to process it, so they send an OK again. > > > > The OK receive, Transmit ACK occurs 4 times. > > A short while later, Vitelity hangs up on my cell phone. > > > > Asterisk is never told the call is gone. > > > > If I hangup the call from Asterisk side, > > Asterisk sends the BYE message. > > Vitelity responds with a ?SIP/2.0 481 Call leg/transaction does not exist? > > > > Again, the trace indicates the ACK message is missing the Contact header. > > > > Additional note: the network admin is asking why the local_net, > external_media_address, and external_signalling_address are needed. He > wrote me??You should NOT have to know your public IP. The firewall > should be doing fixup commands to change the values in the packet? > >First... "The firewall should be doing fixup commands to change the values in the packet? *The firewall should NOT be changing values in the packet. If it is, all bets are off.* Second. Can you try making a call from a phone instead of from an AMI originate?> > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Monday, December 15, 2014 11:14 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP configuration question > > > > On Mon, Dec 15, 2014 at 9:48 PM, Dan Cropp <dan at amtelco.com> wrote: > > Thanks George. > > > > I will correct my local_net in the morning. > > > > Vitelity chan_sip settings I have working, do not have a fromuser. > > sip.conf settings... > > > > I think you can actually specify anything, it just has to be populated > with something other than a sub-account username. > > > > > > [HVout] > > type=friend > > dtmfmode=auto > > host=64.2.142.93 > > disallow=all > > allow=ulaw > > canreinvite=no > > trustrpid=yes > > sendrpid=yes > > nat=yes > > context=TestApp > > > > > On Dec 15, 2014, at 9:32 PM, George Joseph <george.joseph at fairview5.com> > wrote: > > > > > > On Mon, Dec 15, 2014 at 7:34 PM, Dan Cropp <dan at amtelco.com> wrote: > > I am not sure if I entered the correct settings for the transport > information. > > For the local_net, I entered my local ip address, but no mask. I will > check with the network admin so he can verify the settings I entered. > > > > You need the network and mask. For example if the ip address and mask of > the test machine is 192.168.0.1/255.255.255.0 then the correct entry > would be 192.168.0.0/24. > > > > One minor detail, we are using ip authentication. When Vitelity changed > my account from user based authentication to IP based authentication, they > stopped including a user for the account. > > > > Should these settings work without the from_user (IP based authentication) > or do I need to get the account name from Vitelity? > > > > You definitely need the master account login username. If you has this > working with chan_sip, then try the 'fromuser' from sip.conf and user is > from_user. > > > > > > > > > > Have a great day! > > > > Da > > > > *From:* asterisk-users-bounces at lists.digium.com [mailto: > asterisk-users-bounces at lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Monday, December 15, 2014 7:27 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] PJSIP configuration question > > > > Ok Dan, try this... I was able to get this to work behind a NAT and with > ip address authentication. > > [global] > type = global > debug = yes > > [transport1] > type = transport > bind = 0.0.0.0 > protocol = udp > > > > *local_net=<yourlocalnet I.E. 10.10.10.10/24 > <http://10.10.10.10/24>>external_media_address=<your public ip > address>external_signaling_address=<your public address>* > [outbound.vitelity.net] > type = aor > remove_existing = yes > qualify_frequency = 60 > contact = sip:64.2.142.93 > > [outbound.vitelity.net] > type = endpoint > context = TestApp > transport = transport1 > aors = outbound.vitelity.net > dtmf_mode = rfc4733 > force_rport = yes > rtp_symmetric = yes > rewrite_contact = yes > send_rpid = yes > trust_id_inbound = yes > disallow = all > allow = ulaw > direct_media = no > > *from_user=<your main vitelity account name> ; Not subaccount* > > [outbound.vitelity.net] > type = identify > endpoint = outbound.vitelity.net > match = 64.2.142.93 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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/20141216/7dca38fd/attachment-0001.html>
Dan Cropp wrote:> I corrected my local_net setting (based on advice from network admin). > > I have tried several different values for the from_user and still have > the same problem. > > Asterisk receives the OK from Vitelity. > > Asterisk sends the ACK (without a Contact header).A Contact header is not required to be in the ACK.> > Vitelity doesn?t seem to process it, so they send an OK again.I'd try to isolate this further as there's two possible things: 1. The ACK never got to them 2. They didn't process it> > The OK receive, Transmit ACK occurs 4 times. > > A short while later, Vitelity hangs up on my cell phone. > > Asterisk is never told the call is gone. > > If I hangup the call from Asterisk side, > > Asterisk sends the BYE message. > > Vitelity responds with a ?SIP/2.0 481 Call leg/transaction does not exist? > > Again, the trace indicates the ACK message is missing the Contact header. > > Additional note: the network admin is asking why the local_net, > external_media_address, and external_signalling_address are needed. He > wrote me??You should NOT have to know your public IP. The firewall > should be doing fixup commands to change the values in the packet?This can cause major problems. I've rarely (if ever) come across an ALG (that's what that is) that didn't break something. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org