research2*CLI> research2*CLI> research2*CLI> sip set debug on SIP Debugging enabled <--- SIP read from UDP:192.168.51.253:5060 ---> INVITE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK17fe941e Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Tue, 26 Apr 2011 06:51:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "1690" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 312 v=0 o=root 226075433 226075433 IN IP4 192.168.51.253 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.253 t=0 0 m=audio 16870 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 14 lines) --- Sending to 192.168.51.253:5060 (no NAT) Using INVITE request as basis request - 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 Found peer 'gs-0001' for 'gs-0001' from 192.168.51.253:5060 <--- Reliably Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK17fe941e;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as1ca8405f Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20fc76e4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.51.253:5060 ---> ACK sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK17fe941e Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as1ca8405f Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.51.253:5060 ---> INVITE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK0e9445be Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3.2 Authorization: Digest username="gs-0001", realm="asterisk", algorithm=MD5, uri="sip:1689@192.168.51.88:5060", nonce="20fc76e4", response="0a54da254b4375c1cba6fbdbe0cf79b8" Date: Tue, 26 Apr 2011 06:51:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Remote-Party-ID: "1690" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 312 v=0 o=root 226075433 226075434 IN IP4 192.168.51.253 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.253 t=0 0 m=audio 16870 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (16 headers 14 lines) --- Sending to 192.168.51.253:5060 (no NAT) Using INVITE request as basis request - 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 Found peer 'gs-0001' for 'gs-0001' from 192.168.51.253:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format GSM for ID 3 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.51.253:16870 Looking for 1689 in default (domain 192.168.51.88:5060) list_route: hop: <--- Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK0e9445be;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [1689@default:1] Dial("SIP/gs-0001-0000002f", "SIP/1689,15") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800000000000 (testlaw) to SDP Reliably Transmitting (no NAT) to 192.168.51.235:5060: INVITE sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70398110 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Tue, 26 Apr 2011 06:51:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 254 v=0 o=root 815248444 815248444 IN IP4 192.168.51.88 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.88 t=0 0 m=audio 11032 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 1689 Retransmitting #1 (no NAT) to 192.168.51.235:5060: INVITE sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70398110 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Date: Tue, 26 Apr 2011 06:51:23 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 254 v=0 o=root 815248444 815248444 IN IP4 192.168.51.88 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.88 t=0 0 m=audio 11032 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.51.235:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70398110 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: hold Contact: 1689 Server: OpenStage_80_V2 R1.28.0 SIP 110311 X-Siemens-Call-Type: ST-insecure Content-Length: 0 <-------------> --- (12 headers 0 lines) --- -- SIP/1689-00000030 is ringing <--- Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK0e9445be;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP:192.168.51.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70398110 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: hold Contact: 1689 Server: OpenStage_80_V2 R1.28.0 SIP 110311 Supported: replaces, timer X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 1319631969 IN IP4 192.168.51.235 s=SIP Call c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (14 headers 10 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.51.235:5010 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.235:5060 Transmitting (no NAT) to 192.168.51.235:5060: ACK sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK4976b2b2 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- -- SIP/1689-00000030 answered SIP/gs-0001-0000002f Audio is at 5060 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK0e9445be;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 312 v=0 o=root 1613251926 1613251926 IN IP4 192.168.51.88 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.88 t=0 0 m=audio 11076 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Remotely bridging SIP/gs-0001-0000002f and SIP/1689-00000030 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.235:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.51.235:5060: INVITE sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK5fd53c51 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 233 v=0 o=root 815248444 815248445 IN IP4 192.168.51.253 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.253 t=0 0 m=audio 16870 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.51.253:5060 ---> ACK sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK36e7fa15 Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.253:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.51.253:5060: INVITE sip:gs-0001@192.168.51.253:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70fe5e49 Max-Forwards: 70 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 290 v=0 o=root 1613251926 1613251927 IN IP4 192.168.51.235 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.51.253:5060 ---> INVITE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK7686adb4 Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "1690" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 288 v=0 o=root 226075433 226075435 IN IP4 192.168.51.236 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.236 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- <--- Reliably Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK7686adb4;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 104 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <------------> <--- SIP read from UDP:192.168.51.253:5060 ---> SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70fe5e49;received=192.168.51.88 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 102 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.253:5060 Transmitting (no NAT) to 192.168.51.253:5060: ACK sip:gs-0001@192.168.51.253:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK70fe5e49 Max-Forwards: 70 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- [Apr 26 08:51:25] WARNING[11391]: chan_sip.c:19184 handle_response_invite: just did sched_add waitid(65835) for sip_reinvite_retry for dialog 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 in handle_response_invite <--- SIP read from UDP:192.168.51.253:5060 ---> ACK sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK7686adb4 Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sent RTP P2P packet to 192.168.51.253:16870 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.235:5010 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.51.235:5060: INVITE sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK5fd53c51 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 233 v=0 o=root 815248444 815248445 IN IP4 192.168.51.253 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.253 t=0 0 m=audio 16870 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Sent RTP P2P packet to 192.168.51.235:5010 (type 08, len 000160) Sent RTP P2P packet to 192.168.51.235:5010 (type 08, len 000160) <--- SIP read from UDP:192.168.51.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK5fd53c51 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 103 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: hold Contact: 1689 Server: OpenStage_80_V2 R1.28.0 SIP 110311 Supported: replaces, timer X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 1319631969 IN IP4 192.168.51.235 s=SIP Call c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (14 headers 10 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.235:5060 Transmitting (no NAT) to 192.168.51.235:5060: ACK sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK24c426f1 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- <--- SIP read from UDP:192.168.51.253:5060 ---> INVITE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK423f35d8 Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 105 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "1690" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 288 v=0 o=root 226075433 226075436 IN IP4 192.168.51.236 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.236 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- Sending to 192.168.51.253:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.51.236:5010 <--- Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK423f35d8;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 105 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK423f35d8;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 105 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1613251926 1613251928 IN IP4 192.168.51.235 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.51.253:5060 ---> ACK sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK498a15ff Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 105 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sent RTP P2P packet to 192.168.51.235:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.235:5010 (type 00, len 000160) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.253:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.51.253:5060: INVITE sip:gs-0001@192.168.51.253:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK67a9b1b8 Max-Forwards: 70 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 290 v=0 o=root 1613251926 1613251929 IN IP4 192.168.51.235 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.51.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK67a9b1b8;received=192.168.51.88 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.51.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK67a9b1b8;received=192.168.51.88 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 288 v=0 o=root 226075433 226075437 IN IP4 192.168.51.236 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.236 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (12 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.51.236:5010 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.253:5060 Transmitting (no NAT) to 192.168.51.253:5060: ACK sip:gs-0001@192.168.51.253:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK60e93483 Max-Forwards: 70 From: ;tag=as74815fca To: "1690" ;tag=as23257cc5 Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- <--- SIP read from UDP:192.168.51.253:5060 ---> INVITE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK72d79b17 Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 106 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "1690" ;party=called;privacy=off;screen=no Content-Type: application/sdp Content-Length: 288 v=0 o=root 226075433 226075438 IN IP4 192.168.51.236 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.236 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> -- (15 headers 13 lines) --- Sending to 192.168.51.253:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.51.236:5010 <--- Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK72d79b17;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 106 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK72d79b17;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 106 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1613251926 1613251930 IN IP4 192.168.51.235 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.51.253:5060 ---> ACK sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK6e4d777b Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 106 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.51.253:5060 ---> INVITE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK2585e0da Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 107 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Remote-Party-ID: "1690" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 289 v=0 o=root 226075433 226075439 IN IP4 192.168.51.253 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.253 t=0 0 m=audio 16870 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (15 headers 13 lines) --- Sending to 192.168.51.253:5060 (no NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.51.253:16870 <--- Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK2585e0da;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 107 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Length: 0 <------------> Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK2585e0da;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 107 INVITE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 290 v=0 o=root 1613251926 1613251931 IN IP4 192.168.51.235 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from UDP:192.168.51.253:5060 ---> ACK sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK239b12ee Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Contact: Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 107 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.51.253:5060 ---> BYE sip:1689@192.168.51.88:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK5923f0ad Max-Forwards: 70 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 108 BYE User-Agent: Asterisk PBX 1.8.3.2 Authorization: Digest username="gs-0001", realm="asterisk", algorithm=MD5, uri="sip:1689@192.168.51.88:5060", nonce="20fc76e4", response="670037e6250933e3328d0dd978f4680a" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.51.253:5060 (no NAT) Scheduling destruction of SIP dialog '483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.51.253:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK5923f0ad;received=192.168.51.253 From: "1690" ;tag=as23257cc5 To: ;tag=as74815fca Call-ID: 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 CSeq: 108 BYE Server: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 <------------> set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.235:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.51.235:5060: INVITE sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK766db844 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 104 INVITE User-Agent: Asterisk PBX 1.8.3.2 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 231 v=0 o=root 815248444 815248446 IN IP4 192.168.51.88 s=Asterisk PBX 1.8.3.2 c=IN IP4 192.168.51.88 t=0 0 m=audio 11032 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from UDP:192.168.51.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK766db844 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 104 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: hold Contact: 1689 Server: OpenStage_80_V2 R1.28.0 SIP 110311 Supported: replaces, timer X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 1319631969 IN IP4 192.168.51.235 s=SIP Call c=IN IP4 192.168.51.235 t=0 0 m=audio 5010 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=sendrecv <-------------> --- (14 headers 10 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.235:5060 Transmitting (no NAT) to 192.168.51.235:5060: ACK sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK15733fc0 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Contact: Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 104 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- Scheduling destruction of SIP dialog '2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060' in 6400 ms (Method: INVITE) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.235:5060 Reliably Transmitting (no NAT) to 192.168.51.235:5060: BYE sip:1689@192.168.51.235:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK3830b357 Max-Forwards: 70 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 105 BYE User-Agent: Asterisk PBX 1.8.3.2 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- == Spawn extension (default, 1689, 1) exited non-zero on 'SIP/gs-0001-0000002f' <--- SIP read from UDP:192.168.51.235:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.88:5060;branch=z9hG4bK3830b357 From: "1690" ;tag=as6b7420d2 To: ;tag=2385077650 Call-ID: 2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060 CSeq: 105 BYE Server: OpenStage_80_V2 R1.28.0 SIP 110311 X-Siemens-RTP-Stats: PS=7,OS=1120,PR=0,OR=0,PL=0,JI=0,LA=0,SS=0,EN=0,DE=4294967295,IE=00,TCLW=52 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '2a9256e31b540cce57f298976e2e84a0@192.168.51.88:5060' Method: INVITE Really destroying SIP dialog '483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060' Method: BYE research2*CLI> sip set debug off SIP Debugging Disabled research2*CLI>