research1*CLI> research1*CLI> sip set debug on SIP Debugging enabled <--- SIP read from UDP:192.168.51.236:5060 ---> INVITE sip:1689@192.168.51.253:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bKeaa4c1ba7f1d9f577 Max-Forwards: 70 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: Call-ID: cbfebc76443c3d5a CSeq: 1306899281 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: talk, hold Contact: 1690 Min-SE: 90 Supported: timer, replaces, 100rel User-Agent: OpenStage_60_V2 R1.16.0 SIP 100704 X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 293 v=0 o=MxSIP 0 1854943961 IN IP4 192.168.51.236 s=SIP Call c=IN IP4 192.168.51.236 t=0 0 m=audio 5010 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 <-------------> --- (16 headers 13 lines) --- Sending to 192.168.51.236:5060 (no NAT) Using INVITE request as basis request - cbfebc76443c3d5a Found peer '1690' for '1690' from 192.168.51.236:5060 <--- Reliably Transmitting (no NAT) to 192.168.51.236:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bKeaa4c1ba7f1d9f577;received=192.168.51.236 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as7431059a Call-ID: cbfebc76443c3d5a CSeq: 1306899281 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="3b5999d9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'cbfebc76443c3d5a' in 6400 ms (Method: INVITE) <--- SIP read from UDP:192.168.51.236:5060 ---> ACK sip:1689@192.168.51.253:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bKeaa4c1ba7f1d9f577 Max-Forwards: 70 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as7431059a Call-ID: cbfebc76443c3d5a CSeq: 1306899281 ACK User-Agent: OpenStage_60_V2 R1.16.0 SIP 100704 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from UDP:192.168.51.236:5060 ---> INVITE sip:1689@192.168.51.253:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bK04aec4dad198eb133 Max-Forwards: 70 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: Call-ID: cbfebc76443c3d5a CSeq: 1306899282 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: talk, hold Authorization: Digest username="1690",realm="asterisk",nonce="3b5999d9",uri="sip:1689@192.168.51.253:5060;transport=udp",response="f8bd84c8d6270bcb757b1694f883fa05",algorithm=MD5 Contact: 1690 Min-SE: 90 Supported: timer, replaces, 100rel User-Agent: OpenStage_60_V2 R1.16.0 SIP 100704 X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 293 v=0 o=MxSIP 0 1854943961 IN IP4 192.168.51.236 s=SIP Call c=IN IP4 192.168.51.236 t=0 0 m=audio 5010 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=silenceSupp:off - - - - a=fmtp:18 annexb=no a=fmtp:101 0-15 <-------------> --- (17 headers 13 lines) --- Sending to 192.168.51.236:5060 (no NAT) Using INVITE request as basis request - cbfebc76443c3d5a Found peer '1690' for '1690' from 192.168.51.236:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.51.236:5010 Looking for 1689 in default (domain 192.168.51.253:5060) list_route: hop: <--- Transmitting (no NAT) to 192.168.51.236:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bK04aec4dad198eb133;received=192.168.51.236 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: Call-ID: cbfebc76443c3d5a CSeq: 1306899282 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/1690-00000069", "SIP/gs-0002/1689") 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 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.51.88:5060: 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 --- -- Called gs-0002/1689 <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (11 headers 0 lines) --- Transmitting (no NAT) to 192.168.51.88:5060: 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 --- 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 Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.51.88:5060: 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 --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (11 headers 0 lines) --- -- SIP/gs-0002-0000006a is ringing <--- Transmitting (no NAT) to 192.168.51.236:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bK04aec4dad198eb133;received=192.168.51.236 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as518e88b4 Call-ID: cbfebc76443c3d5a CSeq: 1306899282 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:192.168.51.88:5060 ---> 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: ontent-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 <-------------> --- (12 headers 14 lines) --- Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format GSM for ID 3 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=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 192.168.51.88:11076 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 Transmitting (no NAT) to 192.168.51.88:5060: 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 --- -- SIP/gs-0002-0000006a answered SIP/1690-00000069 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP <--- Reliably Transmitting (no NAT) to 192.168.51.236:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bK04aec4dad198eb133;received=192.168.51.236 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as518e88b4 Call-ID: cbfebc76443c3d5a CSeq: 1306899282 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: 233 v=0 o=root 159833314 159833314 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 13360 RTP/AVP 0 8 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Remotely bridging SIP/1690-00000069 and SIP/gs-0002-0000006a set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 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 192.168.51.88:5060: 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 --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (14 headers 13 lines) --- <--- Reliably Transmitting (no NAT) to 192.168.51.88:5060 ---> 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 <------------> <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (12 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 Transmitting (no NAT) to 192.168.51.88:5060: 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 --- [Apr 26 08:51:25] WARNING[3431]: chan_sip.c:19184 handle_response_invite: just did sched_add waitid(65684) for sip_reinvite_retry for dialog 483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060 in handle_response_invite <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (10 headers 0 lines) --- <--- SIP read from UDP:192.168.51.236:5060 ---> ACK sip:1689@192.168.51.253:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bKb09a84c3c9f0bf63c Max-Forwards: 70 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as518e88b4 Call-ID: cbfebc76443c3d5a CSeq: 1306899282 ACK Authorization: Digest username="1690",realm="asterisk",nonce="3b5999d9",uri="sip:1689@192.168.51.253:5060;transport=udp",response="f8bd84c8d6270bcb757b1694f883fa05",algorithm=MD5 User-Agent: OpenStage_60_V2 R1.16.0 SIP 100704 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.236:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.51.236:5060: INVITE sip:1690@192.168.51.236:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK79bdcf05 Max-Forwards: 70 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Contact: Call-ID: cbfebc76443c3d5a 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: 231 v=0 o=root 159833314 159833315 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 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 192.168.51.88:11076 (type 08, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.88:11076 (type 08, len 000160) Retransmitting #1 (no NAT) to 192.168.51.236:5060: INVITE sip:1690@192.168.51.236:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK79bdcf05 Max-Forwards: 70 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Contact: Call-ID: cbfebc76443c3d5a 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: 231 v=0 o=root 159833314 159833315 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 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 192.168.51.88:11076 (type 08, len 000160) <--- SIP read from UDP:192.168.51.236:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK79bdcf05 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Call-ID: cbfebc76443c3d5a CSeq: 102 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: talk, hold Contact: 1690 Server: OpenStage_60_V2 R1.16.0 SIP 100704 Supported: timer, replaces X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 1854943962 IN IP4 192.168.51.236 s=SIP Call c=IN IP4 192.168.51.236 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 192.168.51.236:5010 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.236:5060 Transmitting (no NAT) to 192.168.51.236:5060: ACK sip:1690@192.168.51.236:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK7d6f06ed Max-Forwards: 70 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Contact: Call-ID: cbfebc76443c3d5a CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 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 192.168.51.88:5060: 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 --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (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 192.168.51.235:5010 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 Transmitting (no NAT) to 192.168.51.88:5060: 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 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.236:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Reliably Transmitting (no NAT) to 192.168.51.236:5060: INVITE sip:1690@192.168.51.236:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK14b11857 Max-Forwards: 70 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Contact: Call-ID: cbfebc76443c3d5a 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: 232 v=0 o=root 159833314 159833316 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 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Retransmitting #1 (no NAT) to 192.168.51.236:5060: INVITE sip:1690@192.168.51.236:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK14b11857 Max-Forwards: 70 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Contact: Call-ID: cbfebc76443c3d5a 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: 232 v=0 o=root 159833314 159833316 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 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) <--- SIP read from UDP:192.168.51.236:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK14b11857 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Call-ID: cbfebc76443c3d5a CSeq: 103 INVITE Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,MESSAGE,UPDATE Allow-Events: talk, hold Contact: 1690 Server: OpenStage_60_V2 R1.16.0 SIP 100704 Supported: timer, replaces X-Siemens-Call-Type: ST-insecure Content-Type: application/sdp Content-Length: 202 v=0 o=MxSIP 0 1854943962 IN IP4 192.168.51.236 s=SIP Call c=IN IP4 192.168.51.236 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 192.168.51.236:5060 Transmitting (no NAT) to 192.168.51.236:5060: ACK sip:1690@192.168.51.236:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.51.253:5060;branch=z9hG4bK474b1be1 Max-Forwards: 70 From: ;tag=as518e88b4 To: 1690 ;tag=0ce7d01e45;epid=SC034447 Contact: Call-ID: cbfebc76443c3d5a CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.3.2 Content-Length: 0 --- Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (14 headers 13 lines) --- Sending to 192.168.51.88:5060 (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 192.168.51.235:5010 <--- Transmitting (no NAT) to 192.168.51.88:5060 ---> 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 <------------> 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 192.168.51.88:5060 ---> 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 <------------> <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (10 headers 0 lines) --- Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 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 192.168.51.88:5060: 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 --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (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 192.168.51.235:5010 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 Transmitting (no NAT) to 192.168.51.88:5060: 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 --- Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) Sent RTP P2P packet to 192.168.51.236:5010 (type 00, len 000160) <--- SIP read from UDP:192.168.51.236:5060 ---> BYE sip:1689@192.168.51.253:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bKba9ebe81fe626e96f Max-Forwards: 70 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as518e88b4 Call-ID: cbfebc76443c3d5a CSeq: 1306899283 BYE Authorization: Digest username="1690",realm="asterisk",nonce="3b5999d9",uri="sip:1689@192.168.51.253:5060",response="fa865af38ad0172cc35dd9c7fe163761",algorithm=MD5 Supported: timer User-Agent: OpenStage_60_V2 R1.16.0 SIP 100704 X-Siemens-RTP-Stats: PS=160,OS=25600,PR=0,OR=0,PL=0,JI=0,LA=0,SS=0,EN=0,DE=4294967295 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Sending to 192.168.51.236:5060 (no NAT) Scheduling destruction of SIP dialog 'cbfebc76443c3d5a' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.51.236:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.51.236;branch=z9hG4bKba9ebe81fe626e96f;received=192.168.51.236 From: 1690 ;tag=0ce7d01e45;epid=SC034447 To: ;tag=as518e88b4 Call-ID: cbfebc76443c3d5a CSeq: 1306899283 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 192.168.51.88:5060 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 192.168.51.88:5060: 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 --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (11 headers 0 lines) --- <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (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 192.168.51.235:5010 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 Transmitting (no NAT) to 192.168.51.88:5060: 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 --- Scheduling destruction of SIP dialog '483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.51.88:5060 Reliably Transmitting (no NAT) to 192.168.51.88:5060: 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 --- == Spawn extension (default, 1689, 1) exited non-zero on 'SIP/1690-00000069' <--- SIP read from UDP:192.168.51.88:5060 ---> 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 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '483c15353a2edac6740a1fa705d1bb46@192.168.51.253:5060' Method: ACK research1*CLI> sip set debug off SIP Debugging Disabled research1*CLI>