--- SIP read from UDP://*.*.*.139:5060 ---> INVITE sip:001805123120@*.*.*.10 SIP/2.0 Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-8bda1bc9 From: 1012 ;tag=5651d9715bf3fdfbo0 To: Remote-Party-ID: 1012 ;screen=yes;party=calling Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 101 INVITE Max-Forwards: 70 Contact: 1012 Expires: 240 User-Agent: nst1012 Content-Length: 440 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 48697 48697 IN IP4 *.*.*.139 s=- c=IN IP4 *.*.*.139 t=0 0 m=audio 16472 RTP/AVP 0 8 2 4 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (15 headers 20 lines) --- == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Sending to *.*.*.139 : 5060 (no NAT) Using INVITE request as basis request - 19fc5f29-259873a3@*.*.*.139 Found peer 'nst1012' for 'nst1012' from *.*.*.139:5060 <--- Reliably Transmitting (no NAT) to *.*.*.139:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-8bda1bc9;received=*.*.*.139 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as22aa5131 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 101 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="763b84e4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '19fc5f29-259873a3@*.*.*.139' in 32000 ms (Method: INVITE) sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> ACK sip:001805123120@*.*.*.10 SIP/2.0 Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-8bda1bc9 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as22aa5131 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 101 ACK Max-Forwards: 70 Contact: 1012 User-Agent: nst1012 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> INVITE sip:001805123120@*.*.*.10 SIP/2.0 Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-3e648ef9 From: 1012 ;tag=5651d9715bf3fdfbo0 To: Remote-Party-ID: 1012 ;screen=yes;party=calling Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE Max-Forwards: 70 Authorization: Digest username="nst1012",realm="asterisk",nonce="763b84e4",uri="sip:001805123120@*.*.*.10",algorithm=MD5,response="a54d363b35049e74bcab322d6636599e" Contact: 1012 Expires: 240 User-Agent: nst1012 Content-Length: 440 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura, replaces Content-Type: application/sdp v=0 o=- 48697 48697 IN IP4 *.*.*.139 s=- c=IN IP4 *.*.*.139 t=0 0 m=audio 16472 RTP/AVP 0 8 2 4 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv <-------------> --- (16 headers 20 lines) --- Sending to *.*.*.139 : 5060 (no NAT) Using INVITE request as basis request - 19fc5f29-259873a3@*.*.*.139 Found peer 'nst1012' for 'nst1012' from *.*.*.139:5060 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port *.*.*.139:16472 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format G726-32 for ID 2 Found audio description format G723 for ID 4 Found audio description format G729a for ID 18 Found unknown media description format G726-40 for ID 96 Found unknown media description format G726-24 for ID 97 Found unknown media description format G726-16 for ID 98 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port *.*.*.139:16472 Looking for 001805123120 in callout (domain *.*.*.10) list_route: hop: <--- Transmitting (no NAT) to *.*.*.139:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-3e648ef9;received=*.*.*.139 From: 1012 ;tag=5651d9715bf3fdfbo0 To: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [001805123120@callout:1] Dial("SIP/nst1012-082423d0", "sip/cs1000e/001805123120") in new stack == Using SIP RTP CoS mark 5 == Using UDPTL CoS mark 5 Audio is at *.*.*.10 port 11700 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to *.*.*.200:5060: INVITE sip:001805123120@*.*.*.200;user=phone SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK225697f3;rport Max-Forwards: 70 From: "nst1012" ;tag=as792c55fe To: Contact: Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.6 Date: Fri, 11 Sep 2009 07:50:43 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 259 v=0 o=root 1496728500 1496728500 IN IP4 *.*.*.10 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.10 t=0 0 m=audio 11700 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called cs1000e/001805123120 sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 100 Trying From: "nst1012";tag=as792c55fe To: Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 102 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK225697f3 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 183 Session Progress From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 102 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK225697f3 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 225 v=0 o=- 113230 1 IN IP4 *.*.*.200 s=- t=0 0 m=audio 5448 RTP/AVP 0 101 111 c=IN IP4 *.*.*.143 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- ound RTP audio format 0 Found RTP audio format 101 Found RTP audio format 111 Peer audio RTP is at port *.*.*.143:5448 Found audio description format telephone-event for ID 101 Found unknown media description format X-nt-inforeq for ID 111 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port *.*.*.143:5448 -- SIP/cs1000e-08242cb8 is making progress passing it to SIP/nst1012-082423d0 Audio is at *.*.*.10 port 15034 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (no NAT) to *.*.*.139:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-3e648ef9;received=*.*.*.139 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 1301031912 1301031912 IN IP4 *.*.*.10 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.10 t=0 0 m=audio 15034 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 180 Ringing From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 102 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK225697f3 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 225 v=0 o=- 113230 1 IN IP4 *.*.*.200 s=- t=0 0 m=audio 5448 RTP/AVP 0 101 111 c=IN IP4 *.*.*.143 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv <-------------> --- (12 headers 12 lines) --- -- SIP/cs1000e-08242cb8 is ringing <--- Transmitting (no NAT) to *.*.*.139:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-3e648ef9;received=*.*.*.139 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> -- SIP/cs1000e-08242cb8 is making progress passing it to SIP/nst1012-082423d0 sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 200 OK From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 102 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK225697f3 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 P-Asserted-Identity: Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 225 v=0 o=- 113230 1 IN IP4 *.*.*.200 s=- t=0 0 m=audio 5448 RTP/AVP 0 101 111 c=IN IP4 *.*.*.143 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.200, port 5060 Transmitting (no NAT) to *.*.*.200:5060: ACK sip:001805123120;phone-context=UnknownUnknown@host.sip:5060;maddr=*.*.*.200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK1d694c81;rport Max-Forwards: 70 From: "nst1012" ;tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Contact: Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.6 Content-Length: 0 --- -- SIP/cs1000e-08242cb8 answered SIP/nst1012-082423d0 Audio is at *.*.*.10 port 15034 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to *.*.*.139:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-3e648ef9;received=*.*.*.139 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 259 v=0 o=root 1301031912 1301031913 IN IP4 *.*.*.10 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.10 t=0 0 m=audio 15034 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/nst1012-082423d0 and SIP/cs1000e-08242cb8 set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.200, port 5060 Audio is at *.*.*.10 port 11700 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to *.*.*.200:5060: INVITE sip:001805123120;phone-context=UnknownUnknown@host.sip:5060;maddr=*.*.*.200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK4891089d;rport Max-Forwards: 70 From: "nst1012" ;tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Contact: Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 263 v=0 o=root 1496728500 1496728501 IN IP4 *.*.*.139 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.139 t=0 0 m=audio 16472 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 100 Trying From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 103 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK4891089d Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> ACK sip:001805123120@*.*.*.10 SIP/2.0 Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-88a187f0 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 ACK Max-Forwards: 70 Authorization: Digest username="nst1012",realm="asterisk",nonce="763b84e4",uri="sip:001805123120@*.*.*.10",algorithm=MD5,response="a54d363b35049e74bcab322d6636599e" Contact: 1012 User-Agent: nst1012 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.139, port 5060 Audio is at *.*.*.10 port 15034 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to *.*.*.139:5060: INVITE sip:nst1012@*.*.*.139:5060 SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK29013116;rport Max-Forwards: 70 From: ;tag=as42a03c88 To: 1012 ;tag=5651d9715bf3fdfbo0 Contact: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 1301031912 1301031914 IN IP4 *.*.*.143 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.143 t=0 0 m=audio 5448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 200 OK From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 103 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK4891089d Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 P-Asserted-Identity: Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 225 v=0 o=- 113230 2 IN IP4 *.*.*.200 s=- t=0 0 m=audio 5448 RTP/AVP 0 101 111 c=IN IP4 *.*.*.143 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:111 X-nt-inforeq/8000 a=ptime:20 a=maxptime:20 a=sendrecv <-------------> --- (14 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Found RTP audio format 111 Peer audio RTP is at port *.*.*.143:5448 Found audio description format telephone-event for ID 101 Found unknown media description format X-nt-inforeq for ID 111 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port *.*.*.143:5448 set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.200, port 5060 Transmitting (no NAT) to *.*.*.200:5060: ACK sip:001805123120;phone-context=UnknownUnknown@host.sip:5060;maddr=*.*.*.200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK42221c13;rport Max-Forwards: 70 From: "nst1012" ;tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Contact: Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1.6 Content-Length: 0 --- sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> SIP/2.0 200 OK To: 1012 ;tag=5651d9715bf3fdfbo0 From: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK29013116 Contact: 1012 Server: *.*.*.10 Content-Length: 278 Content-Type: application/sdp v=0 o=- 49175 49175 IN IP4 *.*.*.139 s=- c=IN IP4 *.*.*.139 t=0 0 m=audio 16472 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=silenceSupp:off - - - - <-------------> --- (10 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port *.*.*.139:16472 Found audio description format PCMU for ID 0 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port *.*.*.139:16472 set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.139, port 5060 Transmitting (no NAT) to *.*.*.139:5060: ACK sip:nst1012@*.*.*.139:5060 SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK1209b32f;rport Max-Forwards: 70 From: ;tag=as42a03c88 To: 1012 ;tag=5651d9715bf3fdfbo0 Contact: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.1.6 Content-Length: 0 --- sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> INVITE sip:1012@*.*.*.10 SIP/2.0 From: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 To: "nst1012";tag=as792c55fe Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 1 INVITE Via: SIP/2.0/UDP *.*.*.200:5060;branch=z9hG4bK-57a60a-566097b6-2d7475d Max-Forwards: 70 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 P-Asserted-Identity: Privacy: none Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: application/sdp Content-Length: 94 v=0 o=- 113230 3 IN IP4 *.*.*.200 s=- t=0 0 m=image 5448 udptl t38 c=IN IP4 *.*.*.143 <-------------> --- (15 headers 6 lines) --- Sending to *.*.*.200 : 5060 (no NAT) Got T.38 offer in SDP in dialog 6950a33d17464932647ef3071a7ea93f@*.*.*.10 Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 6950a33d17464932647ef3071a7ea93f@*.*.*.10 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) sbtest*CLI> <--- Transmitting (no NAT) to *.*.*.200:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP *.*.*.200:5060;branch=z9hG4bK-57a60a-566097b6-2d7475d;received=*.*.*.200 From: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 To: "nst1012";tag=as792c55fe Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.139, port 5060 Reliably Transmitting (no NAT) to *.*.*.139:5060: INVITE sip:nst1012@*.*.*.139:5060 SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK15c474e4;rport Max-Forwards: 70 From: ;tag=as42a03c88 To: 1012 ;tag=5651d9715bf3fdfbo0 Contact: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 234 v=0 o=root 1301031912 1301031915 IN IP4 *.*.*.143 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.143 t=0 0 m=image 4613 udptl t38 a=T38Faxversion:0 a=T38MaxBitRate:2400 a=T38FaxRateManagement:transferredTCF a=T38FaxMaxDatagram:376 --- sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> SIP/2.0 488 Not Acceptable Here To: 1012 ;tag=5651d9715bf3fdfbo0 From: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 103 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK15c474e4 Contact: 1012 Warning: 304 spa "Media type not available" Server: *.*.*.10 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.139, port 5060 Transmitting (no NAT) to *.*.*.139:5060: ACK sip:nst1012@*.*.*.139:5060 SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK15c474e4;rport Max-Forwards: 70 From: ;tag=as42a03c88 To: 1012 ;tag=5651d9715bf3fdfbo0 Contact: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 103 ACK User-Agent: Asterisk PBX 1.6.1.6 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.139, port 5060 Audio is at *.*.*.10 port 15034 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to *.*.*.139:5060: INVITE sip:nst1012@*.*.*.139:5060 SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK18e53c09;rport Max-Forwards: 70 From: ;tag=as42a03c88 To: 1012 ;tag=5651d9715bf3fdfbo0 Contact: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 258 v=0 o=root 1301031912 1301031916 IN IP4 *.*.*.143 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.143 t=0 0 m=audio 5448 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- Reliably Transmitting (no NAT) to *.*.*.200:5060 ---> SIP/2.0 488 Not acceptable here Via: SIP/2.0/UDP *.*.*.200:5060;branch=z9hG4bK-57a60a-566097b6-2d7475d;received=*.*.*.200 From: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 To: "nst1012";tag=as792c55fe Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 1 INVITE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 <------------> sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> ACK sip:1012@*.*.*.10 SIP/2.0 From: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 To: "nst1012";tag=as792c55fe Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 1 ACK Via: SIP/2.0/UDP *.*.*.200:5060;branch=z9hG4bK-57a60a-566097b6-2d7475d Max-Forwards: 70 User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> SIP/2.0 200 OK To: 1012 ;tag=5651d9715bf3fdfbo0 From: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 104 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK18e53c09 Contact: 1012 Server: *.*.*.10 Content-Length: 278 Content-Type: application/sdp v=0 o=- 49807 49807 IN IP4 *.*.*.139 s=- c=IN IP4 *.*.*.139 t=0 0 m=audio 16472 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=fmtp:100 192-193 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=silenceSupp:off - - - - <-------------> --- (10 headers 14 lines) --- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port *.*.*.139:16472 Found audio description format PCMU for ID 0 Found unknown media description format NSE for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port *.*.*.139:16472 set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.139, port 5060 Transmitting (no NAT) to *.*.*.139:5060: ACK sip:nst1012@*.*.*.139:5060 SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK7bff2ada;rport Max-Forwards: 70 From: ;tag=as42a03c88 To: 1012 ;tag=5651d9715bf3fdfbo0 Contact: Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 104 ACK User-Agent: Asterisk PBX 1.6.1.6 Content-Length: 0 --- sbtest*CLI> <--- SIP read from UDP://*.*.*.139:5060 ---> BYE sip:001805123120@*.*.*.10 SIP/2.0 Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-5001be1f From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 103 BYE Max-Forwards: 70 Authorization: Digest username="nst1012",realm="asterisk",nonce="763b84e4",uri="sip:001805123120@*.*.*.10",algorithm=MD5,response="793d52af5409fc8eace81a0b8f0febd1" User-Agent: nst1012 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to *.*.*.139 : 5060 (no NAT) sbtest*CLI> <--- Transmitting (no NAT) to *.*.*.139:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP *.*.*.139:5060;branch=z9hG4bK-5001be1f;received=*.*.*.139 From: 1012 ;tag=5651d9715bf3fdfbo0 To: ;tag=as42a03c88 Call-ID: 19fc5f29-259873a3@*.*.*.139 CSeq: 103 BYE Server: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to *.*.*.200, port 5060 Audio is at *.*.*.10 port 11700 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to *.*.*.200:5060: INVITE sip:001805123120;phone-context=UnknownUnknown@host.sip:5060;maddr=*.*.*.200;transport=udp;user=phone SIP/2.0 Via: SIP/2.0/UDP *.*.*.10:5060;branch=z9hG4bK53441201;rport Max-Forwards: 70 From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Contact: Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.6.1.6 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 259 v=0 o=root 1496728500 1496728502 IN IP4 *.*.*.10 s=Asterisk PBX 1.6.1.6 c=IN IP4 *.*.*.10 t=0 0 m=audio 11700 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Scheduling destruction of SIP dialog '6950a33d17464932647ef3071a7ea93f@*.*.*.10' in 32000 ms (Method: ACK) == Spawn extension (callout, 001805123120, 1) exited non-zero on 'SIP/nst1012-082423d0' sbtest*CLI> <--- SIP read from UDP://*.*.*.200:5060 ---> SIP/2.0 100 Trying From: "nst1012";tag=as792c55fe To: ;tag=20095e00-c802000a-13c4-40030-57a603-2c3a546-57a603 Call-ID: 6950a33d17464932647ef3071a7ea93f@*.*.*.10 CSeq: 104 INVITE Via: SIP/2.0/UDP *.*.*.10:5060;rport=5060;branch=z9hG4bK53441201 Supported: 100rel,x-nortel-sipvc,replaces,timer User-Agent: Nortel CS1000 SIP GW release_5.0 version_sse-5.50.12 Contact: Allow: INVITE,ACK,BYE,REGISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '19fc5f29-259873a3@*.*.*.139' Method: BYE