Re: [OpenSIPStack] Problem with OnOutgoingCallConnected

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

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?
>
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