12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.2.153:5060: OPTIONS sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK4c7f3a9b;rport From: "asterisk" ;tag=as4b72ec11 To: Contact: Call-ID: 43189d973c65ea497f542d0347cd6ee4@192.168.2.85 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 17 Jul 2006 02:17:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK4c7f3a9b;rport From: "asterisk" ;tag=as4b72ec11 To: ;tag=FFA9FB08-1D39B911 CSeq: 102 OPTIONS Call-ID: 43189d973c65ea497f542d0347cd6ee4@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '43189d973c65ea497f542d0347cd6ee4@192.168.2.85' -- Accepting call from '00400000000' to '0399999999' on channel 0/8, span 1 -- Executing Set("Zap/8-1", "CALLERID(num)=0400000000") in new stack -- Executing Dial("Zap/8-1", "Local/247@extensions") in new stack -- Called 247@extensions -- Executing Macro("Local/247@extensions-2f59,2", "stdexten|247|SIP/travisk") in new stack -- Executing GotoIf("Local/247@extensions-2f59,2", "1?2:3") in new stack -- Goto (macro-stdexten,s,2) -- Executing SIPAddHeader("Local/247@extensions-2f59,2", "Alert-Info: External") in new stack -- Executing Dial("Local/247@extensions-2f59,2", "SIP/travisk|30") in new stack We're at 192.168.2.85 port 15220 Video is at 192.168.2.85 port 14436 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x100000 (h263p) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x40000 (h261) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 18 lines Reliably Transmitting (no NAT) to 192.168.2.153:5060: INVITE sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK2d5e674d;rport From: "0400000000" ;tag=as280ac901 To: Contact: Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 17 Jul 2006 02:17:14 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Alert-Info: External Content-Type: application/sdp Content-Length: 421 v=0 o=root 18425 18425 IN IP4 192.168.2.85 s=session c=IN IP4 192.168.2.85 t=0 0 m=audio 15220 RTP/AVP 8 0 18 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - m=video 14436 RTP/AVP 103 34 31 a=rtpmap:103 h263-1998/90000 a=rtpmap:34 H263/90000 a=rtpmap:31 H261/90000 --- -- Called travisk ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK2d5e674d;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 CSeq: 102 INVITE Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Length: 0 --- (9 headers 0 lines)--- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK2d5e674d;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 CSeq: 102 INVITE Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Allow-Events: talk,hold,conference Content-Length: 0 --- (10 headers 0 lines)--- Extension Changed 247 new state Ringing for Notify User tassi Extension Changed 247 new state Ringing for Notify User timb Extension Changed 247 new state Ringing for Notify User stevek -- SIP/travisk-08894728 is ringing -- Local/247@extensions-2f59,1 is ringing -- Accepting call from '' to '0399168252' on channel 0/12, span 1 -- Executing Set("Zap/12-1", "CALLERID(num)=") in new stack -- Executing Dial("Zap/12-1", "Local/252@extensions") in new stack -- Called 252@extensions -- Executing Macro("Local/252@extensions-a410,2", "stdexten|252|SIP/michaelm1") in new stack -- Executing GotoIf("Local/252@extensions-a410,2", "0?2:3") in new stack -- Goto (macro-stdexten,s,3) -- Executing Dial("Local/252@extensions-a410,2", "SIP/michaelm1|30") in new stack -- Called michaelm1 -- SIP/michaelm1-088a39f8 is ringing -- Local/252@extensions-a410,1 is ringing Extension Changed 252 new state Ringing for Notify User tassi Extension Changed 252 new state Ringing for Notify User timb Extension Changed 252 new state Ringing for Notify User andrewk Extension Changed 252 new state Ringing for Notify User stevek Extension Changed 252 new state Ringing for Notify User davidw ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK2d5e674d;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 CSeq: 102 INVITE Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Type: application/sdp Content-Length: 296 v=0 o=- 1153102631 1153102631 IN IP4 192.168.2.153 s=Polycom IP Phone c=IN IP4 192.168.2.153 t=0 0 m=audio 2250 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 103 34 31 a=rtpmap:103 h263-1998/90000 a=rtpmap:34 h263/90000 a=rtpmap:31 h261/90000 --- (11 headers 12 lines)--- Found RTP audio format 8 Found RTP audio format 101 Found RTP video format 103 Found RTP video format 34 Found RTP video format 31 Peer audio RTP is at port 192.168.2.153:2250 Found description format PCMA Found description format telephone-event Found description format h263-1998 Found description format h263 Found description format h261 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x1c0000 (h261|h263|h263p), combined - 0x1c0008 (ala w|h261|h263|h263p) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.153, port 5060 Transmitting (no NAT) to 192.168.2.153:5060: ACK sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK0c02daf3;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 Contact: Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/travisk-08894728 answered Local/247@extensions-2f59,2 -- Local/247@extensions-2f59,1 stopped sounds -- Local/247@extensions-2f59,1 answered Zap/8-1 Extension Changed 247 new state InUse for Notify User tassi Extension Changed 247 new state InUse for Notify User timb Extension Changed 247 new state InUse for Notify User stevek -- SIP/michaelm1-088a39f8 answered Local/252@extensions-a410,2 -- Local/252@extensions-a410,1 stopped sounds -- Local/252@extensions-a410,1 answered Zap/12-1 Extension Changed 252 new state InUse for Notify User tassi Extension Changed 252 new state InUse for Notify User timb Extension Changed 252 new state InUse for Notify User andrewk Extension Changed 252 new state InUse for Notify User stevek Extension Changed 252 new state InUse for Notify User davidw ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: INVITE sip:0400000000@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKf9b4539dAD355F0A From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 CSeq: 1 INVITE Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 183 v=0 o=- 1153102631 1153102632 IN IP4 192.168.2.153 s=Polycom IP Phone c=IN IP4 0.0.0.0 t=0 0 m=audio 2250 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 8 lines)--- Using INVITE request as basis request - 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Sending to 192.168.2.153 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:2250 Peer video RTP is at port 0.0.0.0:65535 Found description format PCMA Found description format telephone-event Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on channel 'Zap/8-1' We're at 192.168.2.85 port 15220 Video is at 192.168.2.85 port 14436 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKf9b4539dAD355F0A;received=192.168.2.153 From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 216 v=0 o=root 18425 18426 IN IP4 192.168.2.85 s=session c=IN IP4 192.168.2.85 t=0 0 m=audio 15220 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: ACK sip:0400000000@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKd6cf42bbE1563F00 From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 CSeq: 1 ACK Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: INVITE sip:252@192.168.2.85:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe9e8f66963084AC6 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: CSeq: 1 INVITE Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1153102634 1153102634 IN IP4 192.168.2.153 s=Polycom IP Phone c=IN IP4 192.168.2.153 t=0 0 a=sendrecv m=audio 2252 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 --- (14 headers 11 lines)--- Using INVITE request as basis request - 35343e27-7dd27dc-4c46cd75@192.168.2.153 Sending to 192.168.2.153 : 5060 (non-NAT) Reliably Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe9e8f66963084AC6;received=192.168.2.153 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: ;tag=as590be56a Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6dcaebe6" Content-Length: 0 --- Scheduling destruction of call '35343e27-7dd27dc-4c46cd75@192.168.2.153' in 15000 ms Found user 'travisk' ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: ACK sip:252@192.168.2.85:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe9e8f66963084AC6 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: ;tag=as590be56a CSeq: 1 ACK Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: INVITE sip:252@192.168.2.85:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe0d7cff8B33344C1 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: CSeq: 2 INVITE Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="travisk", realm="asterisk", nonce="6dcaebe6", uri="sip:252@192.168.2.85:5060;user=phone", response="c199b5d93a 1bd4b89b489d19a2eb8dc3", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1153102634 1153102634 IN IP4 192.168.2.153 s=Polycom IP Phone c=IN IP4 192.168.2.153 t=0 0 a=sendrecv m=audio 2252 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 --- (15 headers 11 lines)--- Using INVITE request as basis request - 35343e27-7dd27dc-4c46cd75@192.168.2.153 Sending to 192.168.2.153 : 5060 (non-NAT) Found user 'travisk' Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.153:2252 Peer video RTP is at port 192.168.2.153:65535 Found description format PCMA Found description format PCMU Found description format G729 Found description format telephone-event Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|al aw|g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 252 in extensions (domain 192.168.2.85) list_route: hop: Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe0d7cff8B33344C1;received=192.168.2.153 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/travisk-b704fa48", "stdexten|252|SIP/michaelm1") in new stack -- Executing GotoIf("SIP/travisk-b704fa48", "0?2:3") in new stack -- Goto (macro-stdexten,s,3) -- Executing Dial("SIP/travisk-b704fa48", "SIP/michaelm1|30") in new stack -- Called michaelm1 -- SIP/michaelm1-089279e0 is ringing Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe0d7cff8B33344C1;received=192.168.2.153 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: ;tag=as4f982db0 Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- Extension Changed 252 new state Ringing for Notify User tassi Extension Changed 252 new state Ringing for Notify User timb Extension Changed 252 new state Ringing for Notify User andrewk Extension Changed 252 new state Ringing for Notify User stevek Extension Changed 252 new state Ringing for Notify User davidw -- Channel 0/12, span 1 got hangup request -- Hungup 'Zap/12-1' -- SIP/michaelm1-089279e0 answered SIP/travisk-b704fa48 We're at 192.168.2.85 port 15028 Video is at 192.168.2.85 port 19506 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x100 (g729) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKe0d7cff8B33344C1;received=192.168.2.153 From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: ;tag=as4f982db0 Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 287 v=0 o=root 18425 18425 IN IP4 192.168.2.85 s=session c=IN IP4 192.168.2.85 t=0 0 m=audio 15028 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/travisk-b704fa48 and SIP/michaelm1-089279e0 Extension Changed 252 new state InUse for Notify User tassi ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: ACK sip:252@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bKfbd6084d7A6D957A From: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 To: ;tag=as4f982db0 CSeq: 2 ACK Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Proxy-Authorization: Digest username="travisk", realm="asterisk", nonce="6dcaebe6", uri="sip:252@192.168.2.85:5060;user=phone", response="c199b5d93a 1bd4b89b489d19a2eb8dc3", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.153, port 5060 We're at 192.168.2.85 port 15028 Video is at 192.168.2.85 port 19506 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 13 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.2.153:5060: INVITE sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK1b8bd8c5;rport From: ;tag=as4f982db0 To: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 Contact: Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY X-asterisk-info: SIP re-invite (RTP bridge) Content-Type: application/sdp Content-Length: 217 v=0 o=root 18425 18426 IN IP4 192.168.2.119 s=session c=IN IP4 192.168.2.119 t=0 0 m=audio 5006 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Extension Changed 252 new state InUse for Notify User timb ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK1b8bd8c5;rport From: ;tag=as4f982db0 To: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 CSeq: 102 INVITE Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Type: application/sdp Content-Length: 189 v=0 o=- 1153102634 1153102635 IN IP4 192.168.2.153 s=Polycom IP Phone c=IN IP4 192.168.2.153 t=0 0 m=audio 2252 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (11 headers 8 lines)--- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.153:2252 Peer video RTP is at port 192.168.2.153:65535 Found description format PCMA Found description format telephone-event Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.153, port 5060 Transmitting (no NAT) to 192.168.2.153:5060: ACK sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK5d02f415;rport From: ;tag=as4f982db0 To: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 Contact: Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Extension Changed 252 new state InUse for Notify User andrewk Extension Changed 252 new state InUse for Notify User stevek Extension Changed 252 new state InUse for Notify User davidw Jul 17 12:17:30 NOTICE[18449]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '192.168.2.151' ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: REFER sip:0400000000@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bK3baaedeb63798DF0 From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 CSeq: 2 REFER Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 --- (12 headers 0 lines)--- Transfer to 252 in extensions Transfer from travisk in extensions -- Stopped music on hold on Zap/8-1 Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bK3baaedeb63798DF0;received=192.168.2.153 From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.153, port 5060 Reliably Transmitting (no NAT) to 192.168.2.153:5060: NOTIFY sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK567e797e;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 Contact: Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 ontent-Length: 14I> SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.153, port 5060 Reliably Transmitting (no NAT) to 192.168.2.153:5060: BYE sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK5e8e41b5;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 Contact: Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.153, port 5060 Reliably Transmitting (no NAT) to 192.168.2.153:5060: BYE sip:travisk@192.168.2.153 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK48fa3542;rport From: ;tag=as4f982db0 To: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 Contact: Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Seq: 103 BYE36*CLI> User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Extension Changed 247 new state Idle for Notify User tassi Extension Changed 247 new state Idle for Notify User timb Extension Changed 247 new state Idle for Notify User stevek ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK567e797e;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 CSeq: 103 NOTIFY Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Length: 0 --- (10 headers 0 lines)--- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: BYE sip:0400000000@192.168.2.85 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bK5925041927468E36 From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 CSeq: 3 BYE Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Max-Forwards: 70 Content-Length: 0 --- (10 headers 0 lines)--- Sending to 192.168.2.153 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.153;branch=z9hG4bK5925041927468E36;received=192.168.2.153 From: ;tag=D1E35D8F-C1955964 To: "0400000000" ;tag=as280ac901 Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK5e8e41b5;rport From: "0400000000" ;tag=as280ac901 To: ;tag=D1E35D8F-C1955964 CSeq: 104 BYE Call-ID: 7390d8f951df9dad07f3151c7e362c1b@192.168.2.85 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Length: 0 --- (9 headers 0 lines)--- -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.153 Destroying call '7390d8f951df9dad07f3151c7e362c1b@192.168.2.85' ASTERISK*CLI> <-- SIP read from 192.168.2.153:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.85:5060;branch=z9hG4bK48fa3542;rport From: ;tag=as4f982db0 To: "Travis Knipe" ;tag=8C162FC2-F46F1BD3 CSeq: 103 BYE Call-ID: 35343e27-7dd27dc-4c46cd75@192.168.2.153 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.6.0039 Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '35343e27-7dd27dc-4c46cd75@192.168.2.153' Jul 17 12:17:46 NOTICE[18449]: chan_sip.c:11244 handle_request: Unknown SIP command 'PUBLISH' from '192.168.2.165' -- Channel 0/8, span 1 got hangup request -- Hungup 'Zap/8-1' Extension Changed 252 new state Idle for Notify User tassi Extension Changed 252 new state Idle for Notify User timb Extension Changed 252 new state Idle for Notify User andrewk Extension Changed 252 new state Idle for Notify User stevek Extension Changed 252 new state Idle for Notify User davidw ASTERISK*CLI>