[May 20 11:12:07] Asterisk SVN-branch-1.8-r319938, Copyright (C) 1999 - 2011 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. ========================================================================= [May 20 11:12:07] Connected to Asterisk SVN-branch-1.8-r319938 currently running on fsdev (pid = 11518) fsdev*CLI> Verbosity is at least 3 fsdev*CLI> sip set debug peer 132 [May 20 11:12:14] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '' failed for '209.191.39.117:5060' - No matching peer found fsdev*CLI> sip set debug peer 322 [May 20 11:12:14] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '' failed for '209.191.39.117:5060' - No matching peer found fsdev*CLI> sip set debug peer 322-eng fsdev*CLI> SIP Debugging Enabled for IP: 209.191.39.117 fsdev*CLI> [May 20 11:12:17] NOTICE[11523]: chan_sip.c:13849 check_auth: Correct auth, but based on stale nonce received from ';tag=5e35c995200173e1o3' fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> NOTIFY sip:64.19.145.13 SIP/2.0 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 10464 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-aca92ef Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> --- (11 headers 0 lines) --- <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-aca92ef;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as6c183cc8 Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 10464 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) fsdev*CLI>  == Using SIP RTP CoS mark 5 fsdev*CLI>  -- Executing [7327049020@from-outside:1] Wait("SIP/mg2-00000000", "1") in new stack fsdev*CLI>  -- Executing [7327049020@from-outside:2] Set("SIP/mg2-00000000", "__INCOMINGCLI=7327049020") in new stack -- Executing [7327049020@from-outside:3] Goto("SIP/mg2-00000000", "from-outside-redir,7327049020,1") in new stack -- Goto (from-outside-redir,7327049020,1) -- Executing [7327049020@from-outside-redir:1] Set("SIP/mg2-00000000", "DIALED_PUBLIC_NUMBER=7327049020") in new stack -- Executing [7327049020@from-outside-redir:2] Set("SIP/mg2-00000000", "DIALED_NUMBER=7327049020") in new stack -- Executing [7327049020@from-outside-redir:3] Set("SIP/mg2-00000000", "status=1") in new stack -- Executing [7327049020@from-outside-redir:4] GotoIf("SIP/mg2-00000000", "1?7") in new stack -- Goto (from-outside-redir,7327049020,7) -- Executing [7327049020@from-outside-redir:7] GotoIf("SIP/mg2-00000000", "0?16") in new stack -- Executing [7327049020@from-outside-redir:8] Set("SIP/mg2-00000000", "MAXCALLS=") in new stack -- Executing [7327049020@from-outside-redir:9] GotoIf("SIP/mg2-00000000", "1?16") in new stack -- Goto (from-outside-redir,7327049020,16) -- Executing [7327049020@from-outside-redir:16] GotoIfTime("SIP/mg2-00000000", "*,*,*,*?from-outside-7327049020-tl-allhours-eng,7327049020,1") in new stack -- Goto (from-outside-7327049020-tl-allhours-eng,7327049020,1) -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:1] Set("SIP/mg2-00000000", "__tenant=eng") in new stack -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:2] Set("SIP/mg2-00000000", "CDR(userfield)=eng") in new stack -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:3] Set("SIP/mg2-00000000", "CDR(accountcode)=eng") in new stack -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:4] Set("SIP/mg2-00000000", "MOH=default-eng") in new stack -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:5] GotoIf("SIP/mg2-00000000", "0?nomoh") in new stack -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:6] Set("SIP/mg2-00000000", "CHANNEL(musicclass)=default-eng") in new stack -- Executing [7327049020@from-outside-7327049020-tl-allhours-eng:7] Macro("SIP/mg2-00000000", "tl-huntlist,engtest-eng") in new stack -- Executing [s@macro-tl-huntlist:1] Set("SIP/mg2-00000000", "CALLERID(name)=7327049020") in new stack -- Executing [s@macro-tl-huntlist:2] Goto("SIP/mg2-00000000", "engtest-eng,s,1") in new stack -- Goto (engtest-eng,s,1) == Channel 'SIP/mg2-00000000' jumping out of macro 'tl-huntlist' -- Executing [s@engtest-eng:1] NoOp("SIP/mg2-00000000", "engtest-eng") in new stack -- Executing [s@engtest-eng:2] Dial("SIP/mg2-00000000", "SIP/175-eng&SIP/322-eng") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/175-eng == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x400 (ilbc) to SDP Reliably Transmitting (no NAT) to 209.191.39.117:5060: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2709cac3 Max-Forwards: 70 From: "7327049020" ;tag=as4113db39 To: Contact: Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Date: Fri, 20 May 2011 15:12:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Type: application/sdp Content-Length: 238 v=0 o=root 471646995 471646995 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319938 c=IN IP4 64.19.145.13 t=0 0 m=audio 13862 RTP/AVP 0 97 a=rtpmap:0 PCMU/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=ptime:20 a=sendrecv --- -- Called SIP/322-eng -- SIP/175-eng-00000001 connected line has changed. Saving it until answer for SIP/mg2-00000000 -- SIP/322-eng-00000002 connected line has changed. Saving it until answer for SIP/mg2-00000000 fsdev*CLI>  -- SIP/175-eng-00000001 is ringing fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "7327049020";tag=as4113db39 To: "Poly_test ENG";tag=EB2153EB-B5997DCE Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2709cac3 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Length: 0 <-------------> --- (10 headers 0 lines) --- fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "7327049020";tag=as4113db39 To: "Poly_test ENG";tag=EB2153EB-B5997DCE Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2709cac3 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Allow-Events: talk,hold,conference Content-Length: 0 <-------------> --- (11 headers 0 lines) --- fsdev*CLI>  -- SIP/322-eng-00000002 is ringing fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "7327049020";tag=as4113db39 To: "Poly_test ENG";tag=EB2153EB-B5997DCE Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2709cac3 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Supported: 100rel Supported: replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Type: application/SDP Content-Length: 165 v=0 o=- 1305904343 1305904343 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 m=audio 51796 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> --- (14 headers 8 lines) --- Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 209.191.39.117:51796 fsdev*CLI> list_route: hop: fsdev*CLI> set_destination: Parsing for address/port to send to fsdev*CLI> set_destination: set destination to 209.191.39.117:5060 fsdev*CLI> Transmitting (no NAT) to 209.191.39.117:5060: ACK sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK0c429e79 Max-Forwards: 70 From: "7327049020" ;tag=as4113db39 To: ;tag=EB2153EB-B5997DCE Contact: Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Content-Length: 0 --- fsdev*CLI>  -- SIP/322-eng-00000002 connected line has changed. Saving it until answer for SIP/mg2-00000000 fsdev*CLI>  -- SIP/322-eng-00000002 answered SIP/mg2-00000000 fsdev*CLI> [May 20 11:12:34] DEBUG[11560]: channel.c:6097 ast_set_owners_and_peers: setting peeraccount to eng for SIP/mg2-00000000 from data on channel SIP/322-eng-00000002 fsdev*CLI>  -- Remotely bridging SIP/mg2-00000000 and SIP/322-eng-00000002 set_destination: Parsing for address/port to send to set_destination: set destination to 209.191.39.117:5060 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (no NAT) to 209.191.39.117:5060: INVITE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK05b5e3bd Max-Forwards: 70 From: "7327049020" ;tag=as4113db39 To: ;tag=EB2153EB-B5997DCE Contact: Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 191 v=0 o=root 471646995 471646996 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.8-r319938 c=IN IP4 64.19.145.7 t=0 0 m=audio 11610 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "7327049020";tag=as4113db39 To: "Poly_test ENG";tag=EB2153EB-B5997DCE Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK05b5e3bd Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER Supported: 100rel Supported: replaces User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Type: application/SDP Content-Length: 165 v=0 o=- 1305904343 1305904344 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 m=audio 51796 RTP/AVP 0 a=sendrecv a=rtpmap:0 PCMU/8000 <-------------> fsdev*CLI> --- (14 headers 8 lines) --- fsdev*CLI> Found RTP audio format 0 Found audio description format PCMU for ID 0 fsdev*CLI> Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) fsdev*CLI> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) fsdev*CLI> Peer audio RTP is at port 209.191.39.117:51796 fsdev*CLI> set_destination: Parsing for address/port to send to fsdev*CLI> set_destination: set destination to 209.191.39.117:5060 fsdev*CLI> Transmitting (no NAT) to 209.191.39.117:5060: ACK sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK5efebcae Max-Forwards: 70 From: "7327049020" ;tag=as4113db39 To: ;tag=EB2153EB-B5997DCE Contact: Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Content-Length: 0 --- fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 1 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKe1cf0949FBB2571C Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Supported: 100rel Supported: replaces Max-Forwards: 70 Allow-Events: talk,hold,conference Content-Type: application/SDP Content-Length: 177 v=0 o=- 1305904343 1305904345 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendonly m=audio 51796 RTP/AVP 0 a=sendonly a=rtpmap:0 PCMU/8000 <-------------> --- (16 headers 9 lines) --- Sending to 209.191.39.117:5060 (no NAT) Found RTP audio format 0 Found audio description format PCMU for ID 0 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 209.191.39.117:51796 fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKe1cf0949FBB2571C;received=209.191.39.117 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> fsdev*CLI> Audio is at 5060 fsdev*CLI> Adding codec 0x4 (ulaw) to SDP fsdev*CLI>  <--- Reliably Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKe1cf0949FBB2571C;received=209.191.39.117 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 191 v=0 o=root 471646995 471646997 IN IP4 64.19.145.7 s=Asterisk PBX SVN-branch-1.8-r319938 c=IN IP4 64.19.145.7 t=0 0 m=audio 11610 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=recvonly <------------> fsdev*CLI>  -- Started music on hold, class 'default-eng', on SIP/mg2-00000000 fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 1 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKdb7cd6d8D2D25DA3 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> fsdev*CLI> --- (12 headers 0 lines) --- fsdev*CLI> [May 20 11:12:36] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI> [May 20 11:12:37] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:312@64.19.145.13;user=phone SIP/2.0 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 1 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKd1299946E93AE481 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Supported: 100rel Supported: replaces Max-Forwards: 70 Allow-Events: talk,hold,conference Content-Type: application/SDP Content-Length: 252 v=0 o=- 1305904345 1305904345 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendrecv m=audio 51798 RTP/AVP 0 110 127 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=30 a=rtpmap:127 telephone-event/8000 <-------------> --- (16 headers 11 lines) --- Sending to 209.191.39.117:5060 (no NAT) Using INVITE request as basis request - a26d1014-df5612bf-b6293942@10.0.15.105 Found peer '322-eng' for '322-eng' from 209.191.39.117:5060 <--- Reliably Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKd1299946E93AE481;received=209.191.39.117 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as33995426 Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 1 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="25ee4fec" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'a26d1014-df5612bf-b6293942@10.0.15.105' in 32000 ms (Method: INVITE) fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:312@64.19.145.13;user=phone SIP/2.0 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as33995426 Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 1 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKd1299946E93AE481 Content-Length: 0 <-------------> --- (7 headers 0 lines) --- fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> INVITE sip:312@64.19.145.13;user=phone SIP/2.0 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 2 INVITE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKa5bed65d214172D0 Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Supported: 100rel Supported: replaces Max-Forwards: 70 Allow-Events: talk,hold,conference Authorization: Digest username="322-eng",realm="asterisk",nonce="25ee4fec",uri="sip:312@64.19.145.13;user=phone",response="f20b41f1df13318f4b114e4c2594be77",algorithm=MD5 Content-Type: application/SDP Content-Length: 252 v=0 o=- 1305904345 1305904345 IN IP4 209.191.39.117 s=Polycom IP Phone c=IN IP4 209.191.39.117 t=0 0 a=sendrecv m=audio 51798 RTP/AVP 0 110 127 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=30 a=rtpmap:127 telephone-event/8000 <-------------> --- (17 headers 11 lines) --- Sending to 209.191.39.117:5060 (no NAT) Using INVITE request as basis request - a26d1014-df5612bf-b6293942@10.0.15.105 Found peer '322-eng' for '322-eng' from 209.191.39.117:5060 == Using SIP RTP CoS mark 5 Found RTP audio format 0 Found RTP audio format 110 Found RTP audio format 127 Found audio description format PCMU for ID 0 Found audio description format iLBC for ID 110 Found audio description format telephone-event for ID 127 Capabilities: us - 0x404 (ulaw|ilbc), peer - audio=0x404 (ulaw|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x404 (ulaw|ilbc) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) Peer audio RTP is at port 209.191.39.117:51798 fsdev*CLI> Looking for 312 in from-inside-eng (domain 64.19.145.13) fsdev*CLI> list_route: hop: fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKa5bed65d214172D0;received=209.191.39.117 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> fsdev*CLI>  -- Executing [312@from-inside-eng:1] Macro("SIP/322-eng-00000003", "tl-set-variables2,from-inside-redir-eng,eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:1] Set("SIP/322-eng-00000003", "__tenant=eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:2] Set("SIP/322-eng-00000003", "CDR(userfield)=eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:3] Set("SIP/322-eng-00000003", "__FROM_INSIDE=1") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:4] Set("SIP/322-eng-00000003", "__MOH=default-eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:5] GotoIf("SIP/322-eng-00000003", "1 ?setmoh") in new stack fsdev*CLI>  -- Goto (macro-tl-set-variables2,s,7) fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:7] Set("SIP/322-eng-00000003", "CHANNEL(musicclass)=default-eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:8] Set("SIP/322-eng-00000003", "CHANNEL(parkinglot)=parkinglot_eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-set-variables2:9] Goto("SIP/322-eng-00000003", "from-inside-redir-eng,312,1") in new stack fsdev*CLI>  -- Goto (from-inside-redir-eng,312,1) fsdev*CLI>  == Channel 'SIP/322-eng-00000003' jumping out of macro 'tl-set-variables2' fsdev*CLI>  -- Executing [312@from-inside-redir-eng:1] Macro("SIP/322-eng-00000003", "tl-userexten,SIP/312-eng,312@default-eng,") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten:1] Set("SIP/322-eng-00000003", "__DIALED_NUMBER=312") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten:2] Set("SIP/322-eng-00000003", "__PICKUPMARK=312-eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten:3] ExecIf("SIP/322-eng-00000003", "0?SIPAddHeader(Alert-Info: <>)") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten:4] GotoIf("SIP/322-eng-00000003", "0?doingringgroup") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten:5] Macro("SIP/322-eng-00000003", "tl-userexten-base,SIP/312-eng,312@default-eng,") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:1] GotoIf("SIP/322-eng-00000003", "1?set_options") in new stack fsdev*CLI>  -- Goto (macro-tl-userexten-base,s,8) fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:8] Set("SIP/322-eng-00000003", "OPTIONS=rtT") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:9] Set("SIP/322-eng-00000003", "__PHONE=SIP/312-eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:10] Set("SIP/322-eng-00000003", "__VM_MBOX=312@default-eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:11] Set("SIP/322-eng-00000003", "THISEXT=TL/eng-312") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:12] Set("SIP/322-eng-00000003", "_CLIMYID=eng-312") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:13] Set("SIP/322-eng-00000003", "THISCHAN=TL/312-eng") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:14] GotoIf("SIP/322-eng-00000003", "0?beenhere") in new stack fsdev*CLI>  -- Executing [s@macro-tl-userexten-base:15] Set("SIP/322-eng-00000003", "_ORIG_EXTEN=312") in new stack -- Executing [s@macro-tl-userexten-base:16] Set("SIP/322-eng-00000003", "_ORIG_EXTEN_USER=TL/eng-312") in new stack -- Executing [s@macro-tl-userexten-base:17] Macro("SIP/322-eng-00000003", "tl-notify") in new stack -- Executing [s@macro-tl-notify:1] Set("SIP/322-eng-00000003", "ADDRESS=") in new stack -- Executing [s@macro-tl-notify:2] UserEvent("SIP/322-eng-00000003", "TlNotify,dialed: 312|callerID: 322|tenant: eng") in new stack -- Executing [s@macro-tl-notify:3] NoOp("SIP/322-eng-00000003", "TL/eng-312") in new stack -- Executing [s@macro-tl-notify:4] GotoIf("SIP/322-eng-00000003", "1?s-exit,1") in new stack -- Goto (macro-tl-notify,s-exit,1) -- Executing [s-exit@macro-tl-notify:1] MacroExit("SIP/322-eng-00000003", "") in new stack -- Executing [s@macro-tl-userexten-base:18] Goto("SIP/322-eng-00000003", "checkformat") in new stack -- Goto (macro-tl-userexten-base,s,20) -- Executing [s@macro-tl-userexten-base:20] GotoIf("SIP/322-eng-00000003", "1?cont1") in new stack -- Goto (macro-tl-userexten-base,s,22) -- Executing [s@macro-tl-userexten-base:22] Set("SIP/322-eng-00000003", "RECORD_CALLEE=") in new stack -- Executing [s@macro-tl-userexten-base:23] Macro("SIP/322-eng-00000003", "tl-set-myvariables") in new stack -- Executing [s@macro-tl-set-myvariables:1] Set("SIP/322-eng-00000003", "MY_CHAN=322-eng-00000003") in new stack -- Executing [s@macro-tl-set-myvariables:2] NoOp("SIP/322-eng-00000003", "THECHANNEL=SIP/322-eng-00000003") in new stack -- Executing [s@macro-tl-set-myvariables:3] Set("SIP/322-eng-00000003", "zap=0") in new stack -- Executing [s@macro-tl-set-myvariables:4] GotoIf("SIP/322-eng-00000003", "1?usechannel") in new stack -- Goto (macro-tl-set-myvariables,s,9) -- Executing [s@macro-tl-set-myvariables:9] Set("SIP/322-eng-00000003", "local=0") in new stack -- Executing [s@macro-tl-set-myvariables:10] GotoIf("SIP/322-eng-00000003", "1?useit") in new stack -- Goto (macro-tl-set-myvariables,s,12) -- Executing [s@macro-tl-set-myvariables:12] Set("SIP/322-eng-00000003", "__MYEXTENSION=322") in new stack -- Executing [s@macro-tl-set-myvariables:13] Set("SIP/322-eng-00000003", "__MYID=eng-322") in new stack -- Executing [s@macro-tl-userexten-base:24] Set("SIP/322-eng-00000003", "RECORD_CALLER=") in new stack -- Executing [s@macro-tl-userexten-base:25] Set("SIP/322-eng-00000003", "VM=1") in new stack -- Executing [s@macro-tl-userexten-base:26] Set("SIP/322-eng-00000003", "VMT0=") in new stack -- Executing [s@macro-tl-userexten-base:27] Set("SIP/322-eng-00000003", "CFNAEXT=") in new stack -- Executing [s@macro-tl-userexten-base:28] Set("SIP/322-eng-00000003", "CFNAAN=") in new stack -- Executing [s@macro-tl-userexten-base:29] GotoIf("SIP/322-eng-00000003", "1?done_checkrecord") in new stack -- Goto (macro-tl-userexten-base,s,47) -- Executing [s@macro-tl-userexten-base:47] NoOp("SIP/322-eng-00000003", "") in new stack -- Executing [s@macro-tl-userexten-base:48] NoOp("SIP/322-eng-00000003", "RECORD_CALLEE=") in new stack -- Executing [s@macro-tl-userexten-base:49] NoOp("SIP/322-eng-00000003", "RECORD_CALLER=") in new stack -- Executing [s@macro-tl-userexten-base:50] NoOp("SIP/322-eng-00000003", "OPTIONS=rtT") in new stack -- Executing [s@macro-tl-userexten-base:51] NoOp("SIP/322-eng-00000003", "TOUCH_MONITOR=") in new stack -- Executing [s@macro-tl-userexten-base:52] GotoIf("SIP/322-eng-00000003", "0?next1") in new stack -- Executing [s@macro-tl-userexten-base:53] Set("SIP/322-eng-00000003", "TIMEOUT=") in new stack -- Executing [s@macro-tl-userexten-base:54] GotoIf("SIP/322-eng-00000003", "0?next1") in new stack -- Executing [s@macro-tl-userexten-base:55] Set("SIP/322-eng-00000003", "TIMEOUT=20") in new stack -- Executing [s@macro-tl-userexten-base:56] Set("SIP/322-eng-00000003", "CDR(userfield)=eng") in new stack -- Executing [s@macro-tl-userexten-base:57] Set("SIP/322-eng-00000003", "SCREEN=") in new stack -- Executing [s@macro-tl-userexten-base:58] Set("SIP/322-eng-00000003", "CONFIRM=") in new stack -- Executing [s@macro-tl-userexten-base:59] GotoIf("SIP/322-eng-00000003", "1?getblock") in new stack -- Goto (macro-tl-userexten-base,s,64) -- Executing [s@macro-tl-userexten-base:64] Set("SIP/322-eng-00000003", "BLOCK=") in new stack -- Executing [s@macro-tl-userexten-base:65] GotoIf("SIP/322-eng-00000003", "1?getrecord") in new stack -- Goto (macro-tl-userexten-base,s,69) -- Executing [s@macro-tl-userexten-base:69] Set("SIP/322-eng-00000003", "RECORD=") in new stack -- Executing [s@macro-tl-userexten-base:70] NoOp("SIP/322-eng-00000003", "calleridnum=322 ") in new stack -- Executing [s@macro-tl-userexten-base:71] GotoIf("SIP/322-eng-00000003", "1?screening") in new stack -- Goto (macro-tl-userexten-base,s,103) -- Executing [s@macro-tl-userexten-base:103] GotoIf("SIP/322-eng-00000003", "1?recording") in new stack -- Goto (macro-tl-userexten-base,s,129) -- Executing [s@macro-tl-userexten-base:129] GotoIf("SIP/322-eng-00000003", "1?forwarding") in new stack -- Goto (macro-tl-userexten-base,s,131) -- Executing [s@macro-tl-userexten-base:131] Set("SIP/322-eng-00000003", "FORWARD=") in new stack -- Executing [s@macro-tl-userexten-base:132] GotoIf("SIP/322-eng-00000003", "1?followmecheck") in new stack -- Goto (macro-tl-userexten-base,s,140) -- Executing [s@macro-tl-userexten-base:140] Set("SIP/322-eng-00000003", "FORWARD=") in new stack -- Executing [s@macro-tl-userexten-base:141] Set("SIP/322-eng-00000003", "__FOLLOWME=0") in new stack -- Executing [s@macro-tl-userexten-base:142] GotoIf("SIP/322-eng-00000003", "1?checkchannel") in new stack -- Goto (macro-tl-userexten-base,s,154) -- Executing [s@macro-tl-userexten-base:154] ChanIsAvail("SIP/322-eng-00000003", "SIP/312-eng") in new stack == Using SIP RTP CoS mark 5 Scheduling destruction of SIP dialog '64a9eb563308b8186a39829413cfd3e2@64.19.145.13:5060' in 32000 ms (Method: INVITE) -- Executing [s@macro-tl-userexten-base:155] GotoIf("SIP/322-eng-00000003", "1?chanavail") in new stack -- Goto (macro-tl-userexten-base,s,157) -- Executing [s@macro-tl-userexten-base:157] GotoIf("SIP/322-eng-00000003", "1?dial") in new stack fsdev*CLI>  -- Goto (macro-tl-userexten-base,s,163) -- Executing [s@macro-tl-userexten-base:163] Dial("SIP/322-eng-00000003", "SIP/312-eng,20,rtT") in new stack == Using SIP RTP CoS mark 5 Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 209.191.39.117:5060: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK4f8f2812;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as13a5a74c To: Contact: Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Date: Fri, 20 May 2011 15:12:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "Poly_test ENG" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 193 v=0 o=root 686987562 686987562 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319938 c=IN IP4 64.19.145.13 t=0 0 m=audio 12384 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- -- Called SIP/312-eng <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKa5bed65d214172D0;received=209.191.39.117 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as719ddc8f Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 100 Trying From: "Poly_test ENG";tag=as13a5a74c To: Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK4f8f2812 Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> fsdev*CLI> --- (8 headers 0 lines) --- fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 180 Ringing From: "Poly_test ENG";tag=as13a5a74c To: ;tag=420ee56fa012b4efi0 Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK4f8f2812 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> fsdev*CLI> --- (9 headers 0 lines) --- -- SIP/312-eng-00000005 is ringing fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKa5bed65d214172D0;received=209.191.39.117 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as719ddc8f Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces C fsdev*CLI> ontact: Content-Length: 0 <------------> fsdev*CLI> [May 20 11:12:38] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI> [May 20 11:12:40] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as13a5a74c To: ;tag=420ee56fa012b4efi0 Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 102 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK4f8f2812 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Supported: replaces Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER,UPDATE Content-Type: application/SDP Content-Length: 214 v=0 o=- 15745299 15745299 IN IP4 209.191.39.117 s=- c=IN IP4 209.191.39.117 t=0 0 m=audio 51800 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> fsdev*CLI> --- (12 headers 11 lines) --- fsdev*CLI> Found RTP audio format 0 fsdev*CLI> Found RTP audio format 101 fsdev*CLI> Found audio description format PCMU for ID 0 fsdev*CLI> Found audio description format telephone-event for ID 101 fsdev*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) fsdev*CLI> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) fsdev*CLI> Peer audio RTP is at port 209.191.39.117:51800 fsdev*CLI> list_route: hop: fsdev*CLI> set_destination: Parsing for address/port to send to fsdev*CLI> set_destination: set destination to 209.191.39.117:5060 fsdev*CLI> Transmitting (NAT) to 209.191.39.117:5060: ACK sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK07bc53dc;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as13a5a74c To: ;tag=420ee56fa012b4efi0 Contact: Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319938 C fsdev*CLI> ontent-Length: 0 --- fsdev*CLI>  -- SIP/312-eng-00000005 answered SIP/322-eng-00000003 fsdev*CLI> Audio is at 5060 fsdev*CLI> Adding codec 0x4 (ulaw) to SDP fsdev*CLI> Adding codec 0x400 (ilbc) to SDP fsdev*CLI>  <--- Reliably Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKa5bed65d214172D0;received=209.191.39.117 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as719ddc8f Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 2 INVITE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Type: application/sdp Content-Length: 241 v=0 o=root 689837946 689837946 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319938 c=IN IP4 64.19.145.13 t=0 0 m=audio 12938 RTP/AVP 0 110 a=rtpmap:0 PCMU/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=30 a=ptime:20 a=sendrecv <------------> fsdev*CLI> [May 20 11:12:41] DEBUG[11561]: channel.c:6097 ast_set_owners_and_peers: setting peeraccount to eng for SIP/322-eng-00000003 from data on channel SIP/312-eng-00000005 fsdev*CLI>  -- Locally bridging SIP/322-eng-00000003 and SIP/312-eng-00000005 fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> ACK sip:312@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as719ddc8f Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 2 ACK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK1db0b5b9ABCE6F0C Contact: Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> fsdev*CLI> --- (12 headers 0 lines) --- fsdev*CLI> [May 20 11:12:44] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> NOTIFY sip:64.19.145.13 SIP/2.0 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 10465 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-e43e93e6 Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> fsdev*CLI> --- (11 headers 0 lines) --- fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-e43e93e6;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as6c183cc8 Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 10465 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> fsdev*CLI> Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> REFER sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 2 REFER Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKf5158d774E6ECDBA Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Refer-To: Referred-By: Content-Length: 0 <-------------> fsdev*CLI> --- (13 headers 0 lines) --- fsdev*CLI> Call 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 got a SIP call transfer from caller: (REFER)! fsdev*CLI> SIP transfer to extension 312@from-inside-eng by 322-eng@64.19.145.13 fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKf5158d774E6ECDBA;received=209.191.39.117 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 2 REFER Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Contact: Content-Length: 0 <------------> fsdev*CLI>  -- Stopped music on hold on SIP/mg2-00000000 fsdev*CLI> set_destination: Parsing for address/port to send to fsdev*CLI> set_destination: set destination to 209.191.39.117:5060 fsdev*CLI> Reliably Transmitting (no NAT) to 209.191.39.117:5060: NOTIFY sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2257a516 Max-Forwards: 70 From: "7327049020";tag=as4113db39 To: "Poly_test ENG";tag=EB2153EB-B5997DCE Contact: Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 104 NOTIFY User-Agent: Asterisk PBX SVN-branch-1.8-r319938 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, INFO, PUBLISH Supported: replaces Content-Length: 16 SIP/2.0 200 OK --- fsdev*CLI>  -- Playing 'beep.gsm' (language 'en') fsdev*CLI> Scheduling destruction of SIP dialog 'a26d1014-df5612bf-b6293942@10.0.15.105' in 32000 ms (Method: ACK) fsdev*CLI> set_destination: Parsing for address/port to send to fsdev*CLI> set_destination: set destination to 209.191.39.117:5060 fsdev*CLI> Reliably Transmitting (no NAT) to 209.191.39.117:5060: BYE sip:322-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf4 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK76a2a0b3 Max-Forwards: 70 From: ;tag=as719ddc8f To: "Poly_test ENG";tag=43D82ACA-4A3FF025 Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 102 BYE User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Proxy-Authorization: Digest username="322-eng", realm="asterisk", algorithm=MD5, uri="64.19.145.13", nonce="", response="eac3218b89666699bb97133fa8966982" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "7327049020";tag=as4113db39 To: "Poly_test ENG";tag=EB2153EB-B5997DCE Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 104 NOTIFY Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK2257a516 Contact: Event: refer;id=2 User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Content-Length: 0 <-------------> fsdev*CLI> --- (11 headers 0 lines) --- fsdev*CLI> SIP Response message for INCOMING dialog NOTIFY arrived fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:7327049020@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 3 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK9f9b7295BE74C8 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Content-Length: 0 <-------------> fsdev*CLI> --- (11 headers 0 lines) --- fsdev*CLI> Sending to 209.191.39.117:5060 (no NAT) Scheduling destruction of SIP dialog '75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060' in 32000 ms (Method: BYE) fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK9f9b7295BE74C8;received=209.191.39.117 From: "Poly_test ENG";tag=EB2153EB-B5997DCE To: "7327049020";tag=as4113db39 Call-ID: 75179a011546ca4f270e27ef62327cf1@64.19.145.13:5060 CSeq: 3 BYE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> -- Locally bridging SIP/mg2-00000000 and SIP/312-eng-00000005 set_destination: Parsing for address/port to send to set_destination: set destination to 209.191.39.117:5060 Audio is at 5060 fsdev*CLI> Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 209.191.39.117:5060: INVITE sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK3f551d41;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as13a5a74c To: ;tag=420ee56fa012b4efi0 Contact: Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Remote-Party-ID: "7327049020" ;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 193 v=0 o=root 686987562 686987563 IN IP4 64.19.145.13 s=Asterisk PBX SVN-branch-1.8-r319938 c=IN IP4 64.19.145.13 t=0 0 m=audio 12384 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- fsdev*CLI>  == Spawn extension (engtest-eng, s, 2) exited non-zero on 'SIP/322-eng-00000003' fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 400 SIP Parser Error : Unexpected '\"', line 9, column 99 From: ;tag=as719ddc8f To: "Poly_test ENG";tag=43D82ACA-4A3FF025 Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 102 BYE Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK76a2a0b3 Max-Forwards: 70 User-Agent: Asterisk PBX SVN-branch-1.8-r319938 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Proxy-Authorization: Digest username="322-eng" realm="asterisk" algorithm=MD5 uri="64.19.145.13", nonce="", response="eac3218b89666699bb97133fa8966982" Content-Length: 0 <-------------> fsdev*CLI> --- (12 headers 0 lines) --- fsdev*CLI> SIP Response message for INCOMING dialog BYE arrived fsdev*CLI>  -- Incoming call: Got SIP response 400 "SIP Parser Error : Unexpected '\"', line 9, column 99" back from 209.191.39.117:5060 fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> SIP/2.0 200 OK From: "Poly_test ENG";tag=as13a5a74c To: ;tag=420ee56fa012b4efi0 Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 103 INVITE Via: SIP/2.0/UDP 64.19.145.13:5060;rport=5060;branch=z9hG4bK3f551d41 Contact: "SPA303 Cisco" Server: Cisco/SPA303-7.4.6 Content-Type: application/SDP Content-Length: 214 v=0 o=- 15745299 15745300 IN IP4 209.191.39.117 s=- c=IN IP4 209.191.39.117 t=0 0 m=audio 51800 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv <-------------> fsdev*CLI> --- (10 headers 11 lines) --- fsdev*CLI> Found RTP audio format 0 fsdev*CLI> Found RTP audio format 101 fsdev*CLI> Found audio description format PCMU for ID 0 fsdev*CLI> Found audio description format telephone-event for ID 101 fsdev*CLI> Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) fsdev*CLI> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1 (telephone-event|), combined - 0x0 (nothing) fsdev*CLI> Peer audio RTP is at port 209.191.39.117:51800 fsdev*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to 209.191.39.117:5060 fsdev*CLI> Transmitting (NAT) to 209.191.39.117:5060: ACK sip:312-eng@209.191.39.117:5060;adtnpxyid-1i2c6kcj=bbecf2 SIP/2.0 Via: SIP/2.0/UDP 64.19.145.13:5060;branch=z9hG4bK7ad81fbc;rport Max-Forwards: 70 From: "Poly_test ENG" ;tag=as13a5a74c To: ;tag=420ee56fa012b4efi0 Contact: Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 103 ACK User-Agent: Asterisk PBX SVN-branch-1.8-r319938 C fsdev*CLI> ontent-Length: 0 --- fsdev*CLI> Really destroying SIP dialog '269f250-0-13c4-62-72b2d122-62' Method: REGISTER fsdev*CLI> [May 20 11:12:48] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI> [May 20 11:12:48] NOTICE[11523]: chan_sip.c:13849 check_auth: Correct auth, but based on stale nonce received from ';tag=5e35c995200173e1o3' fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:322@64.19.145.13:5060 SIP/2.0 From: ;tag=420ee56fa012b4efi0 To: "Poly_test ENG";tag=as13a5a74c Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 101 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-21ba85fd Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Content-Length: 0 <-------------> fsdev*CLI> --- (9 headers 0 lines) --- fsdev*CLI> Sending to 209.191.39.117:5060 (NAT) fsdev*CLI> Scheduling destruction of SIP dialog '47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060' in 32000 ms (Method: BYE) fsdev*CLI>  <--- Transmitting (NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-21ba85fd;received=209.191.39.117;rport=5060 From: ;tag=420ee56fa012b4efi0 To: "Poly_test ENG";tag=as13a5a74c Call-ID: 47c3390723ca0c54552b21ce0ed6fd43@64.19.145.13:5060 CSeq: 101 BYE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> fsdev*CLI>  -- Executing [h@from-inside-redir-eng:1] Hangup("SIP/mg2-00000000", "") in new stack fsdev*CLI>  == Spawn extension (from-inside-redir-eng, h, 1) exited non-zero on 'SIP/mg2-00000000' fsdev*CLI>  == Spawn extension (macro-tl-userexten-base, s, 163) exited non-zero on 'SIP/mg2-00000000' in macro 'tl-userexten-base' fsdev*CLI>  == Spawn extension (macro-tl-userexten, s, 5) exited non-zero on 'SIP/mg2-00000000' in macro 'tl-userexten' fsdev*CLI>  == Spawn extension (from-inside-redir-eng, 312, 1) exited non-zero on 'SIP/mg2-00000000' fsdev*CLI> [May 20 11:12:52] NOTICE[11523]: chan_sip.c:23987 handle_request_register: Registration from '"Ri Jiang" ' failed for '209.191.13.243:26300' - No matching peer found fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> BYE sip:312@64.19.145.13:5060 SIP/2.0 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as719ddc8f Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 3 BYE Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKf3d73313A2F9A236 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.3.1734 Accept-Language: en Max-Forwards: 70 Authorization: Digest username="322-eng",realm="asterisk",nonce="25ee4fec",uri="sip:312@64.19.145.13;user=phone",response="f5b373adc9f462522edeeb3af6d1a58f",algorithm=MD5 Content-Length: 0 <-------------> fsdev*CLI> --- (12 headers 0 lines) --- fsdev*CLI> Sending to 209.191.39.117:5060 (no NAT) fsdev*CLI> Scheduling destruction of SIP dialog 'a26d1014-df5612bf-b6293942@10.0.15.105' in 32000 ms (Method: BYE) fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bKf3d73313A2F9A236;received=209.191.39.117 From: "Poly_test ENG";tag=43D82ACA-4A3FF025 To: ;tag=as719ddc8f Call-ID: a26d1014-df5612bf-b6293942@10.0.15.105 CSeq: 3 BYE Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> fsdev*CLI>  <--- SIP read from UDP:209.191.39.117:5060 ---> NOTIFY sip:64.19.145.13 SIP/2.0 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 10466 NOTIFY Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1489e68a Max-Forwards: 70 User-Agent: Cisco/SPA303-7.4.6 Contact: "SPA303 Cisco" Event: keep-alive Content-Length: 0 <-------------> fsdev*CLI> --- (11 headers 0 lines) --- fsdev*CLI>  <--- Transmitting (no NAT) to 209.191.39.117:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 209.191.39.117:5060;branch=z9hG4bK-1489e68a;received=209.191.39.117 From: "SPA303 Cisco";tag=c7d0e91e95d40f0o0 To: ;tag=as6c183cc8 Call-ID: 5d07fe66-394bec48@10.0.15.101 CSeq: 10466 NOTIFY Server: Asterisk PBX SVN-branch-1.8-r319938 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces Content-Length: 0 <------------> fsdev*CLI> Scheduling destruction of SIP dialog '5d07fe66-394bec48@10.0.15.101' in 32000 ms (Method: NOTIFY) fsdev*CLI> exit [May 20 11:13:02] Executing last minute cleanups