[asterisk-users] Odd one way RTP on SIP to SIP calls

[Julio Arruda]

Just a guess in fact..but..
I'm sure others would love to know how is the NGSS (SST now ?) config 
for this purpose, as well as your sip.conf and etc (one note, you are 
running SN09 or ISN09 ?
Not sure, but this also would help others out there.. :-)



Örn Arnarson wrote:
> Julio,
> 
> It seems you had something going there; I disallowed ISUP messages on
> the SIP-T server and now I have two way audio.
> 
> Thanks a lot for your help!
> 
> Best regards,
> Örn
> 
> On 10/1/07, Örn Arnarson <orn at arnarson.net> wrote:
>> You are right, the remote server is a SIP-T.
>>
>> I haven't had any problems connecting it to regular SIP servers
>> thusfar though. Also like I mentioned, I don't have this one-way RTP
>> problem with an earlier version of Asterisk.
>>
>> Thanks for your reply,
>> Örn
>>
>> On 10/1/07, Julio Arruda <jarruda-asterisk at jarruda.com> wrote:
>>> Is this a SIP connection or a SIP-T one? Not sure (don't have access to
>>> my previous life docs :-), but this seems to be a Session Server Trunks
>>> doing SIP-T, not sure is the configuration you want...Have you tried to
>>> contact their support ?
>>> PS: this "c: application/ISUP;version=ANSI88;base=ANSI88", don't
>>> remember seeing in plain SIP calls, so that is why I suspect is
>>> configured as a SIP-T.
>>>
>>> Örn Arnarson wrote:
>>>> Hi everyone,
>>>>
>>>> I'm having an odd problem with one way RTP on SIP to SIP calls.
>>>> I have two SIP servers, one is an Asterisk and the remote SIP server
>>>> is a Nortel SIP server.
>>>>
>>>> When a call comes to the Nortel server through the PSTN and is routed
>>>> to the Asterisk, audio is fine. Two way RTP and no problems. When a
>>>> SIP client registered on the Nortel server calls the Asterisk, the
>>>> Asterisk doesn't seem to send any RTP.
>>>>
>>>> As far as I can tell, there isn't anything wrong with the call setup.
>>>>
>>>> show core version shows:
>>>> Asterisk 1.4.4 built by mark @ d620 on a i686 running Linux on
>>>> 2007-05-17 06:39:34 UTC
>>>>
>>>> SIP and RTP debugging on Asterisk shows this:
>>>> http://www.arnarson.net/~orn/calldebug.txt
>>>>
>>>> On a Trixbox Asterisk server I have at hand (Asterisk 1.2.18 built by
>>>> root @ build.trixbox.org on a i686 running Linux on 2007-04-25
>>>> 19:59:21 UTC) on the same network (same subnet and physical location)
>>>> as the 1.4.4 this problem does not exist. There is no RTP problem when
>>>> SIP clients registered on Nortel call.
>>>>
>>>> If anyone could help or suggest anything it would be greatly appreciated.
>>>>
>>>> Best regards,
>>>> Örn
>>>> _______________________________________________