Hello Ishfaq, and Isrlgb,
The canreinvite value for UA friend entries are set to no, and for
the OpenSIPS peer entry it's set to yes. I do have esternip and
localnet cid set in sip.conf.
I did not want to start a new email, but part of my problem right now
is that OpenSIPS is in charge of
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I
On Thu, 2013-01-03 at 12:47 -0500, Nick Khamis wrote:
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I
Did you set externip and localnet in your sip conf ?
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Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of
the sip.conf settings that may to some seem redundant or have a
misconception with. I do apologize if this has been discussed time and
time again as I would imagine. If anything, this email would make
google
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Thursday, January 03, 2013 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Moving User Agent
List - Non-Commercial Discussion
Subject: [asterisk-users] Moving User Agent To Remote Location
Hello Everyone,
Before getting into SIP and RTP traces, I wanted to clarify some of the
sip.conf settings that may to some seem redundant or have a misconception
with. I do apologize if this has been
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
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On 01/03/2013 02:23 PM, Markus Weiler wrote:
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
I think he means 10007.
--
On Thu, Jan 3, 2013 at 2:21 PM, Nick Khamis sym...@gmail.com wrote:
[Dec 12 15:35:54] WARNING[1736]: app_dial.c:2198 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
Can you check that the registration is happening correctly? Try `sip show
peers` or `sip show peer
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason Parker
Sent: Thursday, January 03, 2013 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving User Agent
2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving User Agent To Remote Location
On 01/03/2013 02:23 PM, Markus Weiler wrote:
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this
one
Just for grins, run netstat -anp on the call using just Asterisk and then
again with OpenSIPS in the mix. It sounds like OpenSIPS or your RTPproxy is
block the audio channels.
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: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason
Parker
Sent: Thursday, January 03, 2013 2:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Moving User Agent To Remote Location
On 01/03/2013
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