Hi! I've investigated this matter further.
Maybe I've found a bug? whithout t38udptlsupport=yes: from the SIP "handshake": (...) Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 213.218.12.2 : 5060 (NAT) Found peer 'toplink' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 2 Found RTP audio format 96 Jun 27 10:31:33 WARNING[3049241520]: chan_sip.c:4797 process_sdp: Unknown or ignored SDP media type in offer: image 10362 udptl t38 Peer audio RTP is at port 195.2.163.101:10360 (...) Callweaver does not recognize the T.38 offer, but the audio port is at 10360. Sniffing with tcpdump gives me: "10:31:36.832474 IP 91.190.224.66.11232 > mgw2-isw1-fra3.de.toplink- voice.net.10360: UDP, length 172" So it is sent to the right port and the voice connection is working. with t38udptlsupport=yes: from the SIP "handshake": (...) Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 213.218.12.2 : 5060 (NAT) Found peer 'toplink' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 2 Found RTP audio format 96 Got T.38 offer in SDP Peer audio RTP is at port 195.2.163.101:11040 Peer T.38 UDPTL is at port 195.2.163.101:11042 (...) So the Peer audio is at Port 11040 and T.38 at 11042. But sniffing with tcpdump gives me the following: "10:21:10.395872 IP 91.190.224.66.13584 > mgw2-isw1-fra3.de.toplink- voice.net.11042: UDP, length 172" So the voice-RTP stream seems to be sent to the T.38 port of the trunk, and I do not hear anything. Is the problem the trunk or am I lacking some parameter in my config to disable T.38 connection in voice calls? Regards, Matthias Am 26.06.2007 um 15:57 schrieb Matthias Gelbhardt: > Hi! > > Have found out something. When I am deactivating T.38 support > (commenting out t38udptlsupport=yes) the incoming audio works. What > is going on there? Which SIP messages will you need to see what is > going on? > > Regards, > > Matthias > > > Am 26.06.2007 um 12:54 schrieb Matthias Gelbhardt: > >> Hi there, >> >> I am testing callweaver at the moment and have a problem. >> >> On outgoing calls it works perfectly, the calling and called party >> could here each other. >> >> On incoming calls there is no voice. When I use tcpdump, I see only >> RTP packets flowing to the SIP provider, but no packet coming from >> it. On outgoing calls I can see the communication flowing in both >> ways. >> >> The strange thing is, although I have canreinvite=no in my sip.conf, >> I have a "Attempting native bridge of" in my log. >> >> The callweaver itself is directly connected to the internet, the >> phones are connected via a second nic on a private net. >> >> A trixbox on a parallel installation works. >> >> I would be happy to provide you with any information you need to help >> me. >> >> Regards, >> >> Matthias >> _______________________________________________ >> Callweaver-users mailing list >> [email protected] >> http://lists.callweaver.org/mailman/listinfo/callweaver-users >> > > _______________________________________________ > Callweaver-users mailing list > [email protected] > http://lists.callweaver.org/mailman/listinfo/callweaver-users > _______________________________________________ Callweaver-users mailing list [email protected] http://lists.callweaver.org/mailman/listinfo/callweaver-users
