Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

[Ilian J. C. P. <ip...@so...>]

Btw, the mod is in OpenSIPStack. Specifically, OpalOSSConnection.cxx.

Ilian Jeri C. Pinzon wrote:
> Hi Tom,
>
> I've checked in a fix for this issue. Can you test if this works in 
> your scenario?
>
> Thanks.
>
> Regards,
> Ilian
>
> tomach wrote:
>> Hello!
>>
>> Yes it is .
>>
>> I localized another problem. Scenario:
>> On sipserver i have this order of codecs:
>> G.711 A
>> G.729
>>
>> On ATLSIP :
>> G.729
>> G.711 A
>>
>> After SipServer send invite to ATLSIP it responses with OK and in SDP 
>> there are two codecs choosen!!! (should be only one). There is:
>> G. 711 A
>> G. 729
>>
>> In ATLSIP I hear some noices all the time, i belive that codec then 
>> is uncorect (finally rtp is using g.711A I sniffed it with ethereal).
>>
>> Whne I call other way: from ATLSIP to SIPSErver it works correct. 
>> always the first in ATLSIP is choosen.
>>
>> Concluding I belive there are problmes when codecs are set between 
>> sipserver and atlsip. (And it seems that ATLSIP has problems wiht 
>> choosing codec).
>> Scenario2:
>>
>> On sipserver i have this order of codecs:
>> G.711 A
>>
>> On ATLSIP :
>> G.729
>> G.711 A
>>
>> After SipServer send invite to ATLSIP it responses with OK and in SDP 
>> there is only  one codec :
>> G. 711 A
>>
>> But this time everything works correct. G711 A is chooosen and 
>> everythig works correct.
>>
>>
>> Did you notice this problem before?
>>
>> ------------------------------------------------------------------------- 
>>
>> This SF.net email is sponsored by DB2 Express
>> Download DB2 Express C - the FREE version of DB2 express and take
>> control of your XML. No limits. Just data. Click to get it now.
>> http://sourceforge.net/powerbar/db2/
>> _______________________________________________
>> opensipstack-devel mailing list
>> ope...@li...
>> https://lists.sourceforge.net/lists/listinfo/opensipstack-devel
>>
>>
>>   
>
>