Hi Nanang, thanks a lot. Yes, you are right. It has been fixed. I really appreciate it. -Jessica> Date: Wed, 25 Jun 2008 16:46:44 +0700> From: nanang@xxxxxxxxx> To: pjsip at lists.pjsip.org> Subject: Re: Remote does not support RFC 2833> > Hi,> > Please see the comment below..> > > 2008/6/25 vopjessie <jessievop at hotmail.com>:> > ..... (cut) ...> > > 14:40:15.625 pjsua_core.c TX 1065 bytes Request msg INVITE/cseq=26500> > (tdta00DACEC8) to UDP 194.221.62.198:5060:> > INVITE sip:vop_dave at sip.voipstunt.com SIP/2.0> > Via: SIP/2.0/UDP> > 220.231.215.199:5100;rport;branch=z9hG4bKPj8c5d52ab1b0c4b65b48525a79a99f75e> > Max-Forwards: 70> > From: sip:vop_abc at sip.voipstunt.com;tag=3cebd378b77445d5aaa2e5566ea15511> > To: sip:vop_dave at sip.voipstunt.com> > Contact: <sip:vop_abc at 220.231.215.199:5100;transport=UDP>> > Call-ID: 1f39bc82b0cb40c1a521302093f659b1> > CSeq: 26500 INVITE> > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, PUBLISH,> > REFER, MESSAGE, OPTIONS> > Supported: replaces, 100rel, norefersub> > User-Agent: PJSUA v0.8.0/win32> > Content-Type: application/sdp> > Content-Length: 444> >> > v=0> > o=- 3423393612 3423393612 IN IP4 220.231.215.199> > s=pjmedia> > c=IN IP4 220.231.215.199> > t=0 0> > a=X-nat:0> > m=audio 4000 RTP/AVP 0 103 102 117 3 104 8 101> > a=rtcp:4001 IN IP4 220.231.215.199> > a=rtpmap:0 PCMU/8000> > a=rtpmap:103 speex/16000> > a=rtpmap:102 speex/8000> > a=rtpmap:117 iLBC/8000> > a=fmtp:117 mode=20> > a=rtpmap:3 GSM/8000> > a=rtpmap:104 speex/32000> > a=rtpmap:8 PCMA/8000> > a=sendrecv> > a=rtpmap:101 telephone-event/8000> > a=fmtp:101 0-15> >> > --end msg--> > 14:40:19.343 sip_endpoint.c Processing incoming message: Response msg> > 200/INVITE/cseq=26500 (rdata00D79EDC)> > 14:40:19.343 pjsua_core.c RX 673 bytes Response msg 200/INVITE/cseq=26500> > (rdata00D79EDC) from UDP 194.221.62.198:5060:> > SIP/2.0 200 Ok> > Via: SIP/2.0/UDP> > 220.231.215.199:5100;branch=z9hG4bKPj8c5d52ab1b0c4b65b48525a79a99f75e;rport> > From: <sip:vop_abc at sip.voipstunt.com>;tag=3cebd378b77445d5aaa2e5566ea15511> > To: <sip:vop_dave at sip.voipstunt.com>;tag=c41710acc42b10ac485fb3b27468f> > Contact: sip:194.221.62.198:5060> > Call-ID: 1f39bc82b0cb40c1a521302093f659b1> > CSeq: 26500 INVITE> > Server: (Very nice Sip Registrar/Proxy Server)> > Allow: ACKBYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE> > Content-Type: application/sdp> > Content-Length: 157> >> > v=0> > o=CARRIER 1214376014 1214376014 IN IP4 77.72.168.9> > s=SIP Call> > c=IN IP4 77.72.168.9> > t=0 0> > m=audio 11128 RTP/AVP 0> > a=rtpmap:0 PCMU/8000> > a=ptime:20> >> > ... (cut) ...> > It seems to be the proxy that doesn't support RFC2833, no telephony> rtpmap in the received 200 OK answer. As you mentioned previously both> parties are pjsua, so DTMF should be sent/received smoothly (you can> try direct call without proxy to test this). Or if the proxy actually> supports RFC2833 then it may be configured incorrectly to do so.> > Regards,> nanang> > _______________________________________________> Visit our blog: http://blog.pjsip.org> > pjsip mailing list> pjsip at lists.pjsip.org> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org _________________________________________________________________ ?????????????MSN????TA????? http://im.live.cn/emoticons/?ID=18 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20080625/11a6581e/attachment.html
