Asterisk SVN-branch-1.4-r63698M, Copyright (C) 1999 - 2007 Digium, Inc. and others. Created by Mark Spencer Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Parsing /usr/local/asterisk/etc/asterisk/asterisk.conf Parsing /usr/local/asterisk/etc/asterisk/extconfig.conf Connected to Asterisk SVN-branch-1.4-r63698M currently running on devel (pid = 25553) devel*CLI> Verbosity is at least 3 devel*CLI> devel*CLI> <--- SIP read from 66.114.80.25:1071 ---> INVITE sip:111@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK3d8b5cd7 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 Date: Thu, 10 May 2007 22:25:16 GMT CSeq: 101 INVITE User-Agent: CSCO/7 Contact: Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp v=0 o=Cisco-SIPUA 9034 6623 IN IP4 192.168.0.217 s=SIP Call c=IN IP4 192.168.0.217 t=0 0 m=audio 29672 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- Sending to 66.114.80.25 : 1071 (NAT) Using INVITE request as basis request - 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 <--- Reliably Transmitting (NAT) to 66.114.80.25:1071 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK3d8b5cd7;received=66.114.80.25 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: ;tag=as05a7efb8 Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 CSeq: 101 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="acepbx.com", nonce="72a068a0" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217' in 32000 ms (Method: INVITE) Found user '12128121207010001' devel*CLI> <--- SIP read from 66.114.80.25:1071 ---> ACK sip:111@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK3d8b5cd7 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: ;tag=as05a7efb8 Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 Date: Thu, 10 May 2007 22:25:16 GMT CSeq: 101 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- devel*CLI> <--- SIP read from 66.114.80.25:1071 ---> INVITE sip:111@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK4b6981f5 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 Date: Thu, 10 May 2007 22:25:16 GMT CSeq: 102 INVITE User-Agent: CSCO/7 Contact: Proxy-Authorization: Digest username="12128121207010001",realm="acepbx.com",uri="sip:66.114.80.26",response="9d813183c39089495bd52d5915d0256d",nonce="72a068a0",algorithm=MD5 Expires: 180 Content-Type: application/sdp Content-Length: 247 v=0 o=Cisco-SIPUA 9034 6623 IN IP4 192.168.0.217 s=SIP Call c=IN IP4 192.168.0.217 t=0 0 m=audio 29672 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-------------> --- (13 headers 11 lines) --- Sending to 66.114.80.25 : 1071 (NAT) Using INVITE request as basis request - 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 Found user '12128121207010001' 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 192.168.0.217:29672 Found description format PCMU for ID 0 Found description format PCMA for ID 8 devel*CLI> Found description format G729 for ID 18 devel*CLI> Found description format telephone-event for ID 101 devel*CLI> Got unsupported a:fmtp in SDP offer devel*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) devel*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) devel*CLI> Peer audio RTP is at port 192.168.0.217:29672 devel*CLI> Looking for 111 in xyz (domain 66.114.80.26) devel*CLI> list_route: hop: devel*CLI> <--- Transmitting (NAT) to 66.114.80.25:1071 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK4b6981f5;received=66.114.80.25 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 CSeq: 102 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> devel*CLI> -- Executing [111@xyz:1] Set("SIP/12128121207010001-081dd788", "TIMEOUT(absolute)=10800") in new stack devel*CLI> -- Channel will hangup at 2007-05-11 01:25:18 UTC. devel*CLI> -- Executing [111@xyz:2] AGI("SIP/12128121207010001-081dd788", "callprocessing.agi") in new stack devel*CLI> -- Launched AGI Script /usr/local/asterisk/var/lib/asterisk/agi-bin/callprocessing.agi devel*CLI> -- AGI Script Executing Application: (SetCallerPres) Options: (allowed_passed_screen) devel*CLI> -- AGI Script Executing Application: (Set) Options: (CALLERID(name)=xyz main|CALLERID(num)=100) devel*CLI> -- AGI Script Executing Application: (Dial) Options: (SIP/xyz011101|15|) devel*CLI> Audio is at 66.114.80.26 port 46284 devel*CLI> Adding codec 0x4 (ulaw) to SDP devel*CLI> Adding non-codec 0x1 (telephone-event) to SDP devel*CLI> Reliably Transmitting (NAT) to 66.114.80.25:1109: INVITE sip:xyz011101@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK23db59cf;rport From: "xyz main" ;tag=as79e36426 To: Contact: Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 CSeq: 102 INVITE User-Agent: AcePBX Max-Forwards: 70 Date: Thu, 10 May 2007 22:25:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 213 v=0 o=root 25553 25553 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 46284 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- devel*CLI> -- Called xyz011101 devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK23db59cf;rport From: "xyz main" ;tag=as79e36426 To: ;tag=3586B7C0-BD2DEC51 CSeq: 102 INVITE Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Content-Length: 0 <-------------> devel*CLI> --- (9 headers 0 lines) --- devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK23db59cf;rport From: "xyz main" ;tag=as79e36426 To: ;tag=3586B7C0-BD2DEC51 CSeq: 102 INVITE Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (10 headers 0 lines) --- devel*CLI> -- SIP/xyz011101-081e3d18 is ringing <--- Transmitting (NAT) to 66.114.80.25:1071 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK4b6981f5;received=66.114.80.25 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: ;tag=as68907c14 Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 CSeq: 102 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK23db59cf;rport From: "xyz main" ;tag=as79e36426 To: ;tag=3586B7C0-BD2DEC51 CSeq: 102 INVITE Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1178813044 1178813044 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2234 RTP/AVP 0 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (11 headers 9 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.250:2234 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.250:2234 list_route: hop: set_destination: Parsing for address/port to send to devel*CLI> set_destination: set destination to 192.168.0.250, port 5060 devel*CLI> Transmitting (NAT) to 66.114.80.25:1109: ACK sip:xyz011101@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK38a5d5fd;rport From: "xyz main" ;tag=as79e36426 To: ;tag=3586B7C0-BD2DEC51 Contact: Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 CSeq: 102 ACK User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- devel*CLI> -- SIP/xyz011101-081e3d18 answered SIP/12128121207010001-081dd788 devel*CLI> Audio is at 66.114.80.26 port 42744 devel*CLI> Adding codec 0x4 (ulaw) to SDP devel*CLI> Adding non-codec 0x1 (telephone-event) to SDP devel*CLI> <--- Reliably Transmitting (NAT) to 66.114.80.25:1071 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK4b6981f5;received=66.114.80.25 From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: ;tag=as68907c14 Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 CSeq: 102 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=root 25553 25553 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 42744 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> devel*CLI> -- Packet2Packet bridging SIP/12128121207010001-081dd788 and SIP/xyz011101-081e3d18 devel*CLI> <--- SIP read from 66.114.80.25:1071 ---> ACK sip:111@66.114.80.26:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.217:5060;branch=z9hG4bK2f6a4c6e From: "xyz main" ;tag=003094c44b8900084d9e89aa-0d3a3f26 To: ;tag=as68907c14 Call-ID: 003094c4-4b890009-2ede8432-0c55ca51@192.168.0.217 Date: Thu, 10 May 2007 22:25:20 GMT CSeq: 102 ACK User-Agent: CSCO/7 Proxy-Authorization: Digest username="12128121207010001",realm="acepbx.com",uri="sip:66.114.80.26",response="9d813183c39089495bd52d5915d0256d",nonce="72a068a0",algorithm=MD5 Content-Length: 0 <-------------> devel*CLI> --- (10 headers 0 lines) --- devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> INVITE sip:100@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKaf9dd5b583E2E834 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 CSeq: 1 INVITE Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 201 v=0 o=- 1178813044 1178813045 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2234 RTP/AVP 0 101 a=sendonly a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 <-------------> --- (14 headers 9 lines) --- Sending to 66.114.80.25 : 1109 (NAT) Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.250:2234 Found description format PCMU for ID 0 Found description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.250:2234 Audio is at 66.114.80.26 port 46284 devel*CLI> Adding codec 0x4 (ulaw) to SDP devel*CLI> Adding non-codec 0x1 (telephone-event) to SDP devel*CLI> <--- Reliably Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKaf9dd5b583E2E834;received=66.114.80.25 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 CSeq: 1 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=root 25553 25554 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 46284 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=recvonly <------------> devel*CLI> -- Started music on hold, class 'xyz', on channel 'SIP/12128121207010001-081dd788' devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> ACK sip:100@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK4d90154fAB96A5F6 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 CSeq: 1 ACK Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> INVITE sip:700@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK7e7570d94A8E7968 From: "poly work" ;tag=D7107017-7B3CDF5C To: CSeq: 1 INVITE Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1178813047 1178813047 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2236 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 66.114.80.25 : 1109 (NAT) Using INVITE request as basis request - 21afaad3-b06521bd-62299f0a@192.168.0.250 <--- Reliably Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK7e7570d94A8E7968;received=66.114.80.25 From: "poly work" ;tag=D7107017-7B3CDF5C To: ;tag=as31a81439 Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 CSeq: 1 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="acepbx.com", nonce="6bb9cd58" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '21afaad3-b06521bd-62299f0a@192.168.0.250' in 32000 ms (Method: INVITE) Found user 'xyz011101' devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> ACK sip:700@pbxtest.acecape.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK7e7570d94A8E7968 From: "poly work" ;tag=D7107017-7B3CDF5C To: ;tag=as31a81439 CSeq: 1 ACK Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> INVITE sip:700@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK701caede755F0C61 From: "poly work" ;tag=D7107017-7B3CDF5C To: CSeq: 2 INVITE Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="6bb9cd58", uri="sip:700@pbxtest.acecape.com:5060;user=phone", response="9e4fb378084fa13f8fa83b34c9c403a1", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1178813047 1178813047 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2236 RTP/AVP 0 8 18 101 devel*CLI> 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 66.114.80.25 : 1109 (NAT) Using INVITE request as basis request - 21afaad3-b06521bd-62299f0a@192.168.0.250 Found user 'xyz011101' 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 192.168.0.250:2236 Found description format PCMU for ID 0 Found description format PCMA for ID 8 devel*CLI> Found description format G729 for ID 18 devel*CLI> Found description format telephone-event for ID 101 devel*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) devel*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) devel*CLI> Peer audio RTP is at port 192.168.0.250:2236 devel*CLI> Looking for 700 in xyz (domain pbxtest.acecape.com) devel*CLI> list_route: hop: devel*CLI> <--- Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK701caede755F0C61;received=66.114.80.25 From: "poly work" ;tag=D7107017-7B3CDF5C To: Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 CSeq: 2 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 < devel*CLI> ------------> devel*CLI> -- Executing [700@xyz:1] Park("SIP/xyz011101-082013b0", "") in new stack devel*CLI> Audio is at 66.114.80.26 port 40892 devel*CLI> Adding codec 0x4 (ulaw) to SDP devel*CLI> Adding non-codec 0x1 (telephone-event) to SDP devel*CLI> <--- Reliably Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK701caede755F0C61;received=66.114.80.25 From: "poly work" ;tag=D7107017-7B3CDF5C To: ;tag=as2578b71c Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 CSeq: 2 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=root 25553 25553 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 40892 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> ACK sip:700@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK5dfb14c569D7B984 From: "poly work" ;tag=D7107017-7B3CDF5C To: ;tag=as2578b71c CSeq: 2 ACK Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="6bb9cd58", uri="sip:700@pbxtest.acecape.com:5060;user=phone", response="9e4fb378084fa13f8fa83b34c9c403a1", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- devel*CLI> == Parked SIP/xyz011101-082013b0 on -1@parkedcalls. Will timeout back to extension [xyz] s, 1 in 600 seconds -- Added extension '1001' priority 1 to parkedcalls -- Started music on hold, class 'default', on channel 'SIP/xyz011101-082013b0' == Spawn extension (xyz, s, 1) exited KEEPALIVE on 'SIP/xyz011101-082013b0' devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> REFER sip:100@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK6a2759df6FDA22C6 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 CSeq: 2 REFER Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Refer-To: Referred-By: Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Call 23816a2d3fde880f560b05385a64b744@66.114.80.26 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 700@xyz by xyz011101@pbxtest.acecape.com <--- Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK6a2759df6FDA22C6;received=66.114.80.25 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 CSeq: 2 REFER User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Stopped music on hold on SIP/xyz011101-082013b0 -- Stopped music on hold on SIP/12128121207010001-081dd788 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.250, port 5060 Reliably Transmitting (NAT) to 66.114.80.25:1109: NOTIFY sip:xyz011101@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK03e3976c;rport From: "xyz main" ;tag=as79e36426 To: ;tag=3586B7C0-BD2DEC51 Contact: Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 CSeq: 103 NOTIFY User-Agent: AcePBX 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 --- devel*CLI> Scheduling destruction of SIP dialog '21afaad3-b06521bd-62299f0a@192.168.0.250' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to devel*CLI> set_destination: set destination to 192.168.0.250, port 5060 devel*CLI> Reliably Transmitting (NAT) to 66.114.80.25:1109: BYE sip:xyz011101@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK44cde217;rport From: ;tag=as2578b71c To: "poly work" ;tag=D7107017-7B3CDF5C Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 CSeq: 102 BYE User-Agent: AcePBX Max-Forwards: 70 Content-Length: 0 --- devel*CLI> -- Started music on hold, class 'default', on channel 'SIP/12128121207010001-081dd788' devel*CLI> Scheduling destruction of SIP dialog '23816a2d3fde880f560b05385a64b744@66.114.80.26' in 32000 ms (Method: REFER) devel*CLI> == Spawn extension (xyz, 111, 2) exited non-zero on 'SIP/xyz011101-082013b0' devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK03e3976c;rport From: "xyz main" ;tag=as79e36426 To: ;tag=3586B7C0-BD2DEC51 CSeq: 103 NOTIFY Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Content-Length: 0 <-------------> --- (10 headers 0 lines) --- devel*CLI> SIP Response message for INCOMING dialog NOTIFY arrived devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> BYE sip:100@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKe49adee9F7AED5B8 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 CSeq: 3 BYE Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Max-Forwards: 70 Content-Length: 0 <-------------> devel*CLI> --- (10 headers 0 lines) --- devel*CLI> Sending to 66.114.80.25 : 1109 (NAT) devel*CLI> Scheduling destruction of SIP dialog '23816a2d3fde880f560b05385a64b744@66.114.80.26' in 32000 ms (Method: BYE) devel*CLI> <--- Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKe49adee9F7AED5B8;received=66.114.80.25 From: ;tag=3586B7C0-BD2DEC51 To: "xyz main" ;tag=as79e36426 Call-ID: 23816a2d3fde880f560b05385a64b744@66.114.80.26 CSeq: 3 BYE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 66.114.80.26:5060;branch=z9hG4bK44cde217;rport From: ;tag=as2578b71c To: "poly work" ;tag=D7107017-7B3CDF5C CSeq: 102 BYE Call-ID: 21afaad3-b06521bd-62299f0a@192.168.0.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Content-Length: 0 <-------------> devel*CLI> --- (9 headers 0 lines) --- devel*CLI> SIP Response message for INCOMING dialog BYE arrived devel*CLI> Really destroying SIP dialog '21afaad3-b06521bd-62299f0a@192.168.0.250' Method: ACK devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> devel*CLI> show Really destroying SIP dialog '164832a2-8f7c3334-57a0c4b5@192.168.0.250' Method: REGISTER devel*CLI> show parkedcalls devel*CLI> Num Channel (Context Extension Pri ) Timeout 1001 SIP/12128121207010001-081dd788 (xyz s 1 ) 593s 1 parked call. devel*CLI> Really destroying SIP dialog 'ab784484-102ef5c6-b4d330df@192.168.0.250' Method: REGISTER devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> INVITE sip:1001@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK8dc9c6633894EEDA From: "poly work" ;tag=19750871-9B78A1AE To: CSeq: 1 INVITE Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1178813062 1178813062 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2238 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 66.114.80.25 : 1109 (NAT) Using INVITE request as basis request - 2c4dcecd-154df2a7-f28616ac@192.168.0.250 <--- Reliably Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK8dc9c6633894EEDA;received=66.114.80.25 From: "poly work" ;tag=19750871-9B78A1AE To: ;tag=as2caf28ee Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 CSeq: 1 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="acepbx.com", nonce="615fcca1" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '2c4dcecd-154df2a7-f28616ac@192.168.0.250' in 32000 ms (Method: INVITE) Found user 'xyz011101' devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> ACK sip:1001@pbxtest.acecape.com:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK8dc9c6633894EEDA From: "poly work" ;tag=19750871-9B78A1AE To: ;tag=as2caf28ee CSeq: 1 ACK Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> INVITE sip:1001@pbxtest.acecape.com:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKa0cb060621F62AB From: "poly work" ;tag=19750871-9B78A1AE To: CSeq: 2 INVITE Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Supported: 100rel,replaces Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="615fcca1", uri="sip:1001@pbxtest.acecape.com:5060;user=phone", response="937ee7b77c35db6c717badb5497d7b65", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 251 v=0 o=- 1178813062 1178813062 IN IP4 192.168.0.250 s=Polycom IP Phone c=IN IP4 192.168.0.250 t=0 0 m=audio 2238 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 66.114.80.25 : 1109 (NAT) Using INVITE request as basis request - 2c4dcecd-154df2a7-f28616ac@192.168.0.250 Found user 'xyz011101' 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 192.168.0.250:2238 Found description format PCMU for ID 0 Found description format PCMA for ID 8 devel*CLI> Found description format G729 for ID 18 devel*CLI> Found description format telephone-event for ID 101 devel*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw) devel*CLI> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) devel*CLI> Peer audio RTP is at port 192.168.0.250:2238 devel*CLI> Looking for 1001 in xyz (domain pbxtest.acecape.com) devel*CLI> list_route: hop: devel*CLI> <--- Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKa0cb060621F62AB;received=66.114.80.25 From: "poly work" ;tag=19750871-9B78A1AE To: Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 CSeq: 2 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 devel*CLI> <------------> devel*CLI> -- Executing [1001@xyz:1] ParkedCall("SIP/xyz011101-081dc208", "1001") in new stack devel*CLI> Audio is at 66.114.80.26 port 46402 devel*CLI> Adding codec 0x4 (ulaw) to SDP devel*CLI> Adding non-codec 0x1 (telephone-event) to SDP devel*CLI> <--- Reliably Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKa0cb060621F62AB;received=66.114.80.25 From: "poly work" ;tag=19750871-9B78A1AE To: ;tag=as4789c01c Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 CSeq: 2 INVITE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=root 25553 25553 IN IP4 66.114.80.26 s=session c=IN IP4 66.114.80.26 t=0 0 m=audio 46402 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <------------> devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> ACK sip:1001@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bKe5795a6f4E8DCB96 From: "poly work" ;tag=19750871-9B78A1AE To: ;tag=as4789c01c CSeq: 2 ACK Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="615fcca1", uri="sip:1001@pbxtest.acecape.com:5060;user=phone", response="937ee7b77c35db6c717badb5497d7b65", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- devel*CLI> -- Playing 'pbx-invalidpark' (language 'en') devel*CLI> <--- SIP read from 66.114.80.25:1109 ---> BYE sip:1001@66.114.80.26 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK2455a8f9E07E08 From: "poly work" ;tag=19750871-9B78A1AE To: ;tag=as4789c01c CSeq: 3 BYE Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 Contact: User-Agent: PolycomSoundPointIP-SPIP_501-UA/2.1.1.0052 Proxy-Authorization: Digest username="xyz011101", realm="acepbx.com", nonce="615fcca1", uri="sip:1001@pbxtest.acecape.com:5060;user=phone", response="482d2f17e8c14281f6f402fda8899155", algorithm=MD5 Max-Forwards: 70 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 66.114.80.25 : 1109 (NAT) <--- Transmitting (NAT) to 66.114.80.25:1109 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.250;branch=z9hG4bK2455a8f9E07E08;received=66.114.80.25 From: "poly work" ;tag=19750871-9B78A1AE To: ;tag=as4789c01c Call-ID: 2c4dcecd-154df2a7-f28616ac@192.168.0.250 CSeq: 3 BYE User-Agent: AcePBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> devel*CLI> [May 10 18:25:44] WARNING[25720]: res_features.c:1923 park_exec: ast_streamfile of pbx-invalidpark failed on SIP/xyz011101-081dc208 -- Channel SIP/xyz011101-081dc208 tried to talk to nonexistent parked call 1001 == Spawn extension (xyz, 1001, 1) exited non-zero on 'SIP/xyz011101-081dc208' devel*CLI> Really destroying SIP dialog '2c4dcecd-154df2a7-f28616ac@192.168.0.250' Method: BYE