*CLI> sip set debug peer tassi2 SIP Debugging Enabled for IP: 192.168.2.125:5060 *CLI> -- Accepting call from '0263' to '0399999999' on channel 0/8, span 1 -- Executing [0399999999@main:1] Set("Zap/8-1", "CALLERID(num)=263") in new stack -- Executing [0399999999@main:2] Dial("Zap/8-1", "Local/263@extensions") in new stack -- Called 263@extensions -- Executing [263@extensions:1] Macro("Local/263@extensions-5eda,2", "stdexten|263|SIP/tassi&SIP/tassi2") in new stack -- Executing [s@macro-stdexten:1] GotoIf("Local/263@extensions-5eda,2", "0?2:4") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Dial("Local/263@extensions-5eda,2", "SIP/tassi&SIP/tassi2|30") in new stack [Jan 22 17:54:30] WARNING[19729]: app_dial.c:1081 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) Video is at 192.168.2.86 port 11770 Audio is at 192.168.2.86 port 15252 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x40000 (h261) to SDP Adding codec 0x80000 (h263) to SDP Adding codec 0x100000 (h263p) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.2.125:5060: INVITE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK1ccf50ba;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: Contact: Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 22 Jan 2007 06:54:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 420 v=0 o=root 19347 19347 IN IP4 192.168.2.86 s=session c=IN IP4 192.168.2.86 b=CT:384 t=0 0 m=audio 15252 RTP/AVP 8 0 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 11770 RTP/AVP 31 34 103 a=rtpmap:31 H261/90000 a=rtpmap:34 H263/90000 a=rtpmap:103 h263-1998/90000 a=sendrecv --- -- Called tassi2 <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK1ccf50ba;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 CSeq: 102 INVITE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK1ccf50ba;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 CSeq: 102 INVITE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/tassi2-09db4268 is ringing -- Local/263@extensions-5eda,1 is ringing <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK1ccf50ba;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 CSeq: 102 INVITE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Type: application/sdp Content-Length: 320 v=0 o=- 1169448887 1169448887 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2226 RTP/AVP 8 101 a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 31 34 103 a=inactive a=rtpmap:31 h261/90000 a=rtpmap:34 h263/90000 a=rtpmap:103 h263-1998/90000 <-------------> --- (11 headers 14 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found RTP video format 31 Found RTP video format 34 Found RTP video format 103 Peer audio RTP is at port 192.168.2.125:2226 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Found description format h261 for ID 31 Found description format h263 for ID 34 Found description format h263-1998 for ID 103 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x1c0008 (alaw|h261|h263|h263p)/video=0x1c0000 (h261|h263|h263p), combined - 0x1c0008 (alaw|h261|h26 3|h263p) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.125:2226 list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Transmitting (no NAT) to 192.168.2.125:5060: ACK sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK1839282c;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 Contact: Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/tassi2-09db4268 answered Local/263@extensions-5eda,2 -- Local/263@extensions-5eda,1 stopped sounds -- Local/263@extensions-5eda,1 answered Zap/8-1 == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Local/263@extensions-5eda,2' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Local/263@extensions-5eda,2' <--- SIP read from 192.168.2.125:5060 ---> INVITE sip:263@192.168.2.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK8721a4a244750EEF From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e CSeq: 1 INVITE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1169448887 1169448888 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2226 RTP/AVP 8 101 a=sendonly a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 9 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.125:2226 Found description format PCMA for ID 8 Found description format telephone-event for ID 101 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.125:2226 Peer video RTP is at port 192.168.2.125:35840 Audio is at 192.168.2.86 port 15252 Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK8721a4a244750EEF;received=192.168.2.125 From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 19347 19348 IN IP4 192.168.2.86 s=session c=IN IP4 192.168.2.86 t=0 0 m=audio 15252 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on channel 'Zap/8-1' <--- SIP read from 192.168.2.125:5060 ---> ACK sip:263@192.168.2.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK46ad48866BEB63D From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e CSeq: 1 ACK Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- +- <--- SIP read from 192.168.2.125:5060 ---> INVITE sip:254@192.168.2.86:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKc133b99eB2F1F5FB From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: CSeq: 1 INVITE Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1169448894 1169448894 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2228 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 11 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Using INVITE request as basis request - a131ad64-822c81a9-29b2555a@192.168.2.125 <--- Reliably Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKc133b99eB2F1F5FB;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: ;tag=as5677b785 Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="706c42c6" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'a131ad64-822c81a9-29b2555a@192.168.2.125' in 32000 ms (Method: INVITE) Found user 'tassi2' <--- SIP read from 192.168.2.125:5060 ---> ACK sip:254@192.168.2.86:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bKc133b99eB2F1F5FB From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: ;tag=as5677b785 CSeq: 1 ACK Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- SIP read from 192.168.2.125:5060 ---> INVITE sip:254@192.168.2.86:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK210cb6d5C45513D6 From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: CSeq: 2 INVITE Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="tassi2", realm="asterisk", nonce="706c42c6", uri="sip:254@192.168.2.86:5060;user=phone", response="c1d2c9c9c91bb0281b8f459f95468362", algorit hm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1169448894 1169448894 IN IP4 192.168.2.125 s=Polycom IP Phone c=IN IP4 192.168.2.125 t=0 0 m=audio 2228 RTP/AVP 0 8 18 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (15 headers 11 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Using INVITE request as basis request - a131ad64-822c81a9-29b2555a@192.168.2.125 Found user 'tassi2' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.2.125:2228 Found description format PCMU for ID 0 Found description format PCMA for ID 8 Found description format G729 for ID 18 Found description format telephone-event for ID 101 Capabilities: us - 0x1c010e (gsm|ulaw|alaw|g729|h261|h263|h263p), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.2.125:2228 Peer video RTP is at port 192.168.2.125:35840 Looking for 254 in extensions (domain 192.168.2.86) list_route: hop: <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK210cb6d5C45513D6;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [254@extensions:1] Macro("SIP/tassi2-09db1aa8", "stdexten|254|SIP/chrisf1") in new stack -- Executing [s@macro-stdexten:1] GotoIf("SIP/tassi2-09db1aa8", "0?2:4") in new stack -- Goto (macro-stdexten,s,4) -- Executing [s@macro-stdexten:4] Dial("SIP/tassi2-09db1aa8", "SIP/chrisf1|30") in new stack -- Called chrisf1 -- SIP/chrisf1-09da8748 is ringing <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK210cb6d5C45513D6;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: ;tag=as574442ec Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- SIP/chrisf1-09da8748 answered SIP/tassi2-09db1aa8 Audio is at 192.168.2.86 port 18466 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 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK210cb6d5C45513D6;received=192.168.2.125 From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: ;tag=as574442ec Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 312 v=0 o=root 19347 19347 IN IP4 192.168.2.153 s=session c=IN IP4 192.168.2.153 t=0 0 m=audio 2232 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 - - - - a=ptime:20 a=sendrecv <------------> <--- SIP read from 192.168.2.125:5060 ---> ACK sip:254@192.168.2.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK696c5112FFC8E19F From: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 To: ;tag=as574442ec CSeq: 2 ACK Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Proxy-Authorization: Digest username="tassi2", realm="asterisk", nonce="706c42c6", uri="sip:254@192.168.2.86:5060;user=phone", response="c1d2c9c9c91bb0281b8f459f95468362", algorit hm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- No such command '+-' (type 'help' for help) **** At this point SIP/A and SIP/B are connected, but cannot hear eachother. **** <--- SIP read from 192.168.2.125:5060 ---> REFER sip:263@192.168.2.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK484fd47849FA4E6D From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e CSeq: 2 REFER Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 254@extensions by tassi2@192.168.2.86 <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK484fd47849FA4E6D;received=192.168.2.125 From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 2 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on Zap/8-1 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: NOTIFY sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK18cdcaa9;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 Contact: Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 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 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- Scheduling destruction of SIP dialog 'a131ad64-822c81a9-29b2555a@192.168.2.125' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: BYE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK5017a648;rport From: ;tag=as574442ec To: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '6abdbb316bc84f507815eddb216efd1d@192.168.2.86' in 6400 ms (Method: REFER) == Spawn extension (main, 0399999999, 2) exited non-zero on 'SIP/tassi2-09db1aa8' <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK18cdcaa9;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 CSeq: 103 NOTIFY Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived <--- SIP read from 192.168.2.125:5060 ---> BYE sip:263@192.168.2.86 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK271d290e134CFFAB From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e CSeq: 3 BYE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Max-Forwards: 70 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Sending to 192.168.2.125 : 5060 (no NAT) Scheduling destruction of SIP dialog '6abdbb316bc84f507815eddb216efd1d@192.168.2.86' in 6400 ms (Method: BYE) <--- Transmitting (no NAT) to 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.125;branch=z9hG4bK271d290e134CFFAB;received=192.168.2.125 From: ;tag=A787BC6C-13068891 To: "Tass Iliopoulos" ;tag=as18c84d3e Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK5017a648;rport From: ;tag=as574442ec To: "Tass Iliopoulos" ;tag=AA7128C7-8F4A0300 CSeq: 102 BYE Call-ID: a131ad64-822c81a9-29b2555a@192.168.2.125 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog 'a131ad64-822c81a9-29b2555a@192.168.2.125' Method: ACK -- Channel 0/8, span 1 got hangup request == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Zap/8-1' in macro 'stdexten' == Spawn extension (macro-stdexten, s, 4) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: BYE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK45ba8ae0;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 104 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '6abdbb316bc84f507815eddb216efd1d@192.168.2.86' in 6400 ms (Method: BYE) <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 500 Internal Server Error Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK45ba8ae0;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 CSeq: 104 BYE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 Contact: User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived -- Incoming call: Got SIP response 500 "Internal Server Error" back from 192.168.2.125 *CLI> sip set no set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.2.125, port 5060 Reliably Transmitting (no NAT) to 192.168.2.125:5060: BYE sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK21179b71;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Scheduling destruction of SIP dialog '6abdbb316bc84f507815eddb216efd1d@192.168.2.86' in 6400 ms (Method: BYE) <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 481 Call Leg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK21179b71;rport From: "Tass Iliopoulos" ;tag=as18c84d3e To: ;tag=A787BC6C-13068891 CSeq: 105 BYE Call-ID: 6abdbb316bc84f507815eddb216efd1d@192.168.2.86 User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '6abdbb316bc84f507815eddb216efd1d@192.168.2.86' Method: BYE sip no debug No such command 'ssip no' (type 'help' for help) *CLI> Reliably Transmitting (no NAT) to 192.168.2.125:5060: OPTIONS sip:tassi2@192.168.2.125 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK4a3278c1;rport From: "asterisk" ;tag=as47b313db To: Contact: Call-ID: 72a63d37321dc8230bcbbf137b6e36d6@192.168.2.86 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 22 Jan 2007 06:55:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- <--- SIP read from 192.168.2.125:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.86:5060;branch=z9hG4bK4a3278c1;rport From: "asterisk" ;tag=as47b313db To: ;tag=9BE808D9-7D5D77CA CSeq: 102 OPTIONS Call-ID: 72a63d37321dc8230bcbbf137b6e36d6@192.168.2.86 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_430-UA/2.0.3.0127 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog '72a63d37321dc8230bcbbf137b6e36d6@192.168.2.86' Method: OPTIONS *CLI> sip no debug SIP Debugging Disabled