[OpenSER-Users] Changing Audio Ports

[Daniel-Constantin Mierla]

On 08/20/07 21:05, Kelvin Williams wrote:
> THANK YOU!!!!!  
>
> The only problem here is the vendor, Arris Interactive believes their
> products are the greatest.  And "officially do not support SIP Deployment."
>
> Could you point me in the direction of some documentation as to how I can
> drop the 183 and generate a 182 instead?
>   
generating other reply code is a bit hard, needs some coding. Dropping 
183 is easy:
http://openser.org/dokuwiki/doku.php/core-cookbook:devel#drop

Cheers,
Daniel

> Thanks,
> kw
>
> -----Original Message-----
> From: Daniel-Constantin Mierla [mailto:daniel at voice-system.ro] 
> Sent: Monday, August 20, 2007 12:30 PM
> To: kelvin-lists at williamschadwell.com
> Cc: users at openser.org
> Subject: Re: [OpenSER-Users] Changing Audio Ports
>
> Hello,
>
> the sip device is broken, should update the destination port and IP from 
> the 200 ok reply. The best is to ask the vendor to fix it. Think about 
> para;lel forking, there could be many 183 coming from different 
> addresses and ports, what would do the device?
>
> Temporary solution is to drop any 183 or to use perl substitutions for 
> such replies (all to be done in onreply_route).
>
> Cheers,
> Daniel
>
> On 08/18/07 07:24, kelvin-lists at williamschadwell.com wrote:
>   
>> In a previous query I asked if someone could shed some light on as to why
>> my endpoints do not receive any audio when the call is redirected to an
>> announcement server.
>>
>> After a lot of testing, I believe I have found the problem.
>>
>> When I initiate a call from my end point the end point advises the callee
>> as to the port the RTP traffic will be present.  When the call is handed
>> to my PSTN gateway the Gateway responds with its port for RTP in a 183
>> Session Progress.
>>
>> When that call fails (due to timeout) we want to send it over to Asterisk
>> where an announcement will be played to the caller--however the caller
>> never hears it.  The traces show that Asterisk advertises its RTP on a
>> different port that that of the PSTN Gateway.  Some of my endpoints (Cisco
>> IP Phone 7940 and Sipura devices) see and listen for the audio on the new
>> advertised port, however my Arris EMTAs do not, it appears as though they
>> are still "tuned in" to the original audio port advised by the PSTN
>> gateway.
>>
>> My question, is is possible to strip away the "m=audio 22040 RTP/AVP 0 8
>> 18 101." from the SIP message?  I would like to strip it away in the event
>> of a 183 from my gateway (that advertises the port), but pass it when the
>> call is actually answered.
>>
>> If it is not possible to strip the RTP port information away from the
>> message, what would be the best way in handling a situation like this.
>>
>> Many thanks in advance.
>> kw
>>
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>>     
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