We have a SIP trunk between the Telephone operator (O2 CZ) and our ASBCE. When the call gets out to certain numbers and this called party uses the 183 early media option, the calling person does not hear any ring tone and also no voice from the caller. I have made a few traces, where I can see that for both the Early media call and the "standard" call our SBCE offers a G711 A-law codec in the INVITE, then we receive an UPDATE from the operator who adds 101 Telephone event to the RTP map. Then the ASBCE shows in the trace that it is sending the UPDATE to the SM, but the SM never receives it! The operator then repeat sending the UPDATE to our ASBCE, but since we are not responding to the request the call is then terminated.
The SIP Call is as follows:
Skype end-point -> Avaya SM -> Avaya SBCE -> PSTN Operator O2 CZ -> End user's mobile phone.
In the media rules config, we use "Allow Preffered codecs only" and I have listed the PCMA (8) and telephone-event(D) codecs.
We don't do any signalling manipulations or anything...
In the Server Interworking profile for the PSTN I have tried to tinker with the 183 Handling (None, SDP, No SDP) but that didn't made any differenece![[bugeyed]](/data/assets/smilies/bugeyed.gif "[bugeyed] [bugeyed]")
Any Idea what I should look for and how to set this up properly?
I can provide the SBCE/SM SIP traces on request.
regards
Jou
The SIP Call is as follows:
Skype end-point -> Avaya SM -> Avaya SBCE -> PSTN Operator O2 CZ -> End user's mobile phone.
In the media rules config, we use "Allow Preffered codecs only" and I have listed the PCMA (8) and telephone-event(D) codecs.
We don't do any signalling manipulations or anything...
In the Server Interworking profile for the PSTN I have tried to tinker with the 183 Handling (None, SDP, No SDP) but that didn't made any differenece
Any Idea what I should look for and how to set this up properly?
I can provide the SBCE/SM SIP traces on request.
regards
Jou
