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@xxxxxxxxxxxxxxxxx;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@xxxxxxxxxxxxxxxxx>;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
