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 pe
Did you set externip and localnet in your sip conf ?
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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
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
Audio"
>
> Cheers Guys!
>
> Nick
>
> On 1/3/13, Danny Nicholas wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jason
>> Parker
>> Sent: Thursd
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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lf 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 02:23 PM, Markus Weiler wrote:
>> Am 03.01.2013 21:21, schrieb Ni
-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
Assuming this is not a NAT issue (which looks like it), the "no audio"
when signaling works usually means some misconfiguration of codecs
Stelios
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On Thu, Jan 3, 2013 at 2:21 PM, Nick Khamis 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 [destination ph
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.
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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 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 To Remote Location
>
> Hello Everyone,
>
> Before getting into SIP and RTP traces, I wanted to clarif
-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 To
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
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