El Miércoles, 26 de Noviembre de 2008, Fernando Mercês escribió: > Thank you by the interest, Iñaki. > > I don't know the answer to this new INVITE from PBX. In a normal > operation, the RTP traffic should be initiated, right? > > About the RFC, so, the server (voicemail in this case) do not send the > list of acceptable codecs, unfortunately. Tha language (en-us), type > (audio) and codec (G.711 uLaw) are supported for both sides. > > Can you help me to check this new INVITE? I have a .pcap file with > this part of capture.
Could you just paste the SIP trace here? they should be no more than 10 messages. If the SIP traffic goes through a Linux you can use "ngrep" command: ~# ngrep -d any -W byline -T port 5060 -- Iñaki Baz Castillo _______________________________________________ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
