asterisk*CLI> sip debug peer sipfarm SIP Debugging Enabled for IP: 206.80.70.46:5060 -- Executing SetCIDNum("SIP/107-81f6", "+13142664000") in new stack -- Executing Dial("SIP/107-81f6", "SIP/sipfarm/3143212222") in new stack We're at 206.80.66.138 port 60592 Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting: INVITE sip:3143212222@sipfarm.netlogic.net SIP/2.0 Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK0617aa02 From: "Matt S" ;tag=as26a25627 To: Contact: Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 02 Feb 2005 17:49:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 8412 8412 IN IP4 206.80.66.138 s=session c=IN IP4 206.80.66.138 t=0 0 m=audio 60592 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 206.80.70.46:5060 -- Called sipfarm/3143212222 asterisk*CLI> Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK0617aa02 From: "Matt S" ;tag=as26a25627 To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.334a Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sipfarm.netlogic.net", nonce="blah", qop="auth" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 206.80.70.46:5060 "Noisy feedback tells: pid=27326 req_src_ip=206.80.66.138 req_ src_port=5060 in_uri=sip:3143212222@sipfarm.netlogic.net out_uri=sip:3143212222@sipfarm.netlog ic.net via_cnt==1" 10 headers, 0 lines Transmitting: ACK sip:3143212222@sipfarm.netlogic.net SIP/2.0 Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK0617aa02 From: "Matt S" ;tag=as26a25627 To: ;tag=b27e1a1d33761e85846fc98f5f3a7e58.334a Contact: Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 206.80.70.46:5060 We're at 206.80.66.138 port 60592 Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:3143212222@sipfarm.netlogic.net SIP/2.0 Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK1b7f568d -- BELOW -- From: "3143212222" ;tag=as26a25627 -- ABOVE -- To: Contact: Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 103 INVITE User-Agent: Asterisk PBX Proxy-Authorization: Digest username="+18162565163", realm="sipfarm.netlogic.net", algorithm=M D5, uri="sip:3143212222@sipfarm.netlogic.net", nonce="blah ", response="blah2", opaque="", qop="auth", cnonce="301282d3", nc=0 0000001 Date: Wed, 02 Feb 2005 17:49:12 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 218 v=0 o=root 8412 8413 IN IP4 206.80.66.138 s=session c=IN IP4 206.80.66.138 t=0 0 m=audio 60592 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - (no NAT) to 206.80.70.46:5060 asterisk*CLI> Sip read: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK1b7f568d From: "3143212222" ;tag=as26a25627 To: Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 103 INVITE Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 206.80.70.46:5060 "Noisy feedback tells: pid=27325 req_src_ip=206.80.66.138 req_ src_port=5060 in_uri=sip:3143212222@sipfarm.netlogic.net out_uri=sip:+13143212222@209.247.17.5 :5060 via_cnt==1" 9 headers, 0 lines asterisk*CLI> Sip read: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK1b7f568d Record-Route: From: "3143212222" ;tag=as26a25627 To: ;tag=VPST50603522629893 Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 103 INVITE Contact: Content-Type: application/sdp Content-Length: 183 v=0 o=- 1107367433 1107367434 IN IP4 209.247.5.190 s=- c=IN IP4 209.247.5.190 t=0 0 m=audio 60694 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 10 headers, 9 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 209.247.5.190:60694 Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x1 00 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) -- SIP/sipfarm-cabe is making progress passing it to SIP/107-81f6 asterisk*CLI> Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK1b7f568d Record-Route: From: "3143212222" ;tag=as26a25627 To: ;tag=VPST50603522629893 Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 103 INVITE Contact: Content-Type: application/sdp Content-Length: 183 v=0 o=- 1107367433 1107367434 IN IP4 209.247.5.190 s=- c=IN IP4 209.247.5.190 t=0 0 m=audio 60694 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 10 headers, 9 lines Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 209.247.5.190:60694 Found description format telephone-event Capabilities: us - 0x100 (g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723) Transmitting: ACK sip:209.247.17.5:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK3246bca5 Route: From: "Matt S" ;tag=as26a25627 To: ;tag=VPST50603522629893 Contact: Call-ID: 3a909d36504715cf30fc538f64f3a058@206.80.66.138 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 206.80.70.46:5060 -- SIP/sipfarm-cabe answered SIP/107-81f6 -- Attempting native bridge of SIP/107-81f6 and SIP/sipfarm-cabe