voip*CLI> ******************** START HERE No such command '******************** START HERE' (type 'help ******************** START' for other possible commands) voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> <-------------> voip*CLI> <--- SIP read from 203.176.185.10:5060 ---> INVITE sip:61390010834@192.168.0.21:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKa2d1.cb523e7a8c3428042531a8d24c8f6178.0 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5e84da6d6850d73906b203365f3574ce;rport=5061 Max-Forwards: 16 From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 876175924-925917493-929182820-2 h323-conf-id: 876175924-925917493-929182820-2 H323-credit-time: 7200 Content-disposition: session Content-Length: 286 Content-Type: application/sdp v=0 o=Sippy 145902380 0 IN IP4 203.176.185.10 s=VoipCall t=0 0 m=audio 60268 RTP/AVP 18 4 8 0 101 c=IN IP4 203.176.185.10 a=rtpmap:18 g729/8000/1 a=abcde:20 a=rtpmap:4 g723/8000/1 a=rtpmap:8 pcma/8000/1 a=rtpmap:0 pcmu/8000/1 a=rtpmap:101 telephone-event/8000/1 a=sendrecv <-------------> --- (18 headers 13 lines) --- Sending to 203.176.185.10 : 5060 (no NAT) Using INVITE request as basis request - call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o [Mar 9 17:49:53] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5116]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '97891942' Found peer 'worlddialpoint' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 203.176.185.10:60268 Found audio description format g729 for ID 18 Found audio description format g723 for ID 4 Found audio description format pcma for ID 8 Found audio description format pcmu for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80c (ulaw|alaw|g726), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 203.176.185.10:60268 Looking for 61390010834 in incoming (domain 192.168.0.21) [Mar 9 17:49:53] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5116]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '61390010834' AND context = 'did' AND priority = '1' [Mar 9 17:49:53] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5116]: res_config_mysql.c:260 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten LIKE '\\_%' AND context = 'did' AND priority = '1' ORDER BY exten list_route: hop: <--- Transmitting (no NAT) to 203.176.185.10:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKa2d1.cb523e7a8c3428042531a8d24c8f6178.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5e84da6d6850d73906b203365f3574ce;rport=5061 Record-Route: From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '61390010834' AND context = 'did' AND priority = '1' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:260 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten LIKE '\\_%' AND context = 'did' AND priority = '1' ORDER BY exten [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '61390010834' AND context = 'did' AND priority = '1' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:260 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten LIKE '\\_%' AND context = 'did' AND priority = '1' ORDER BY exten [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '61390010834' AND context = 'did' AND priority = '1' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:260 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten LIKE '\\_%' AND context = 'did' AND priority = '1' ORDER BY exten -- Executing [61390010834@incoming:1] Goto("SIP/61390010834-092d8820", "did|0390010834|1") in new stack -- Goto (did,0390010834,1) [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '1' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '1' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '1' -- Executing Macro("SIP/61390010834-092d8820", "setgroupifblank|100001") -- Executing [s@macro-setgroupifblank:1] Set("SIP/61390010834-092d8820", "GROUP()=100001") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:2] NoOp("SIP/61390010834-092d8820", "") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: NoOp -- Executing [s@macro-setgroupifblank:3] Set("SIP/61390010834-092d8820", "_ACCOUNT=100001") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:4] Set("SIP/61390010834-092d8820", "CDR(accountcode)=100001") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:5] Set("SIP/61390010834-092d8820", "BILLSRC=97891942") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:6] Set("SIP/61390010834-092d8820", "BILLDST=0390010834") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:7] Set("SIP/61390010834-092d8820", "CDR(userfield)=incoming,97891942 -> 0390010834") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:8] NoOp("SIP/61390010834-092d8820", "") in new stack [Mar 9 17:49:53] DEBUG[5302]: app_macro.c:373 _macro_exec: Executed application: NoOp [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '2' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '2' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '2' -- Executing Macro("SIP/61390010834-092d8820", "extension|100001|102") -- Executing [s@macro-extension:1] Dial("SIP/61390010834-092d8820", "SIP/100001102|10") in new stack Audio is at 192.168.0.21 port 19710 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.13:5060: INVITE sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK40bfdce3;rport From: "97891942" ;tag=as36804318 To: Contact: Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:49:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 316 v=0 o=root 4964 4964 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19710 RTP/AVP 0 3 8 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 100001102 [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten = '0390010834' AND context = 'did' AND priority = '-1' [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[5302]: res_config_mysql.c:260 realtime_multi_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM extensions_table WHERE exten LIKE '\\_%' AND context = 'did' AND priority = '-1' ORDER BY exten [Mar 9 17:49:53] DEBUG[4972]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[4972]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' AND host = 'dynamic' [Mar 9 17:49:53] DEBUG[4972]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:53] DEBUG[4972]: res_config_mysql.c:140 realtime_mysql: <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK40bfdce3 From: "97891942" ;tag=as36804318 To: Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 102 INVITE Content-Length: 0 <-------------> MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' --- (7 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK40bfdce3 Max-Forwards: 70 From: "97891942" ;tag=as36804318 To: ;tag=xlo563toz4 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/100001102-092dddb8 is ringing voip*CLI> <--- Transmitting (no NAT) to 203.176.185.10:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKa2d1.cb523e7a8c3428042531a8d24c8f6178.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5e84da6d6850d73906b203365f3574ce;rport=5061 Record-Route: From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: ;tag=as476f268a Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21;rport;branch=z9hG4bK40bfdce3 Max-Forwards: 70 From: "97891942" ;tag=as36804318 To: ;tag=xlo563toz4 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 102 INVITE Contact: Accept: application/sdp Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Type: application/sdp Content-Length: 256 v=0 o=100001102 461331375 461331375 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5008 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5009 <-------------> --- (14 headers 13 lines) --- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.13:5008 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.13:5008 [Mar 9 17:49:55] DEBUG[5116]: chan_sip.c:5692 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0xc (ulaw|alaw) and not 0x2 (gsm) list_route: hop: [Mar 9 17:49:55] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.13, port 5060 Transmitting (NAT) to 192.168.0.13:5060: ACK sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK6a47cc58;rport From: "97891942" ;tag=as36804318 To: ;tag=xlo563toz4 Contact: Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/100001102-092dddb8 answered SIP/61390010834-092d8820 Audio is at 192.168.0.21 port 15834 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 203.176.185.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKa2d1.cb523e7a8c3428042531a8d24c8f6178.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5e84da6d6850d73906b203365f3574ce;rport=5061 Record-Route: From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: ;tag=as476f268a Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 200 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 4964 4964 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 15834 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Packet2Packet bridging SIP/61390010834-092d8820 and SIP/100001102-092dddb8 [Mar 9 17:49:55] DEBUG[4972]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:55] DEBUG[4972]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' AND host = 'dynamic' [Mar 9 17:49:55] DEBUG[4972]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:49:55] DEBUG[4972]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' voip*CLI> <--- SIP read from 203.176.185.10:5060 ---> ACK sip:61390010834@192.168.0.21:5060 SIP/2.0 Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKa2d1.8087d23fa58d9f35b73e1367a738a71f.0 Via: SIP/2.0/UDP 203.176.185.10:5061;rport=5061;branch=z9hG4bKb9523c94387cb11f585003d944190034 Max-Forwards: 16 From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: ;tag=as476f268a Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 200 ACK Expires: 300 User-Agent: Sippy <-------------> --- (10 headers 0 lines) --- Really destroying SIP dialog 'e1-200093462545c274416453e19216f' Method: REGISTER voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> INVITE sip:97891942@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKru021.7lfupmbla67tq07yfjxgd Max-Forwards: 70 From: ;tag=xlo563toz4 To: ;tag=as36804318 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 12498 INVITE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Content-Type: application/sdp Content-Length: 339 v=0 o=100001102 461331375 461331376 IN IP4 0.0.0.0 s=- c=IN IP4 0.0.0.0 t=0 0 a=sendonly m=audio 5008 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=rtcp:5009 <-------------> --- (12 headers 17 lines) --- Sending to 192.168.0.13 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:5008 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 0.0.0.0:5008 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKru021.7lfupmbla67tq07yfjxgd;received=192.168.0.13 From: ;tag=xlo563toz4 To: ;tag=as36804318 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 12498 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.0.21 port 19710 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKru021.7lfupmbla67tq07yfjxgd;received=192.168.0.13 From: ;tag=xlo563toz4 To: ;tag=as36804318 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 12498 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 4964 4965 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19710 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> -- Started music on hold, class 'default', on SIP/61390010834-092d8820 voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ACK sip:97891942@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKltwcnhc1hbt3f88xthu Max-Forwards: 70 From: ;tag=xlo563toz4 To: ;tag=as36804318 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 12498 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- [Mar 9 17:49:57] DEBUG[5302]: rtp.c:3222 bridge_p2p_loop: Oooh, formats changed, backing out voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> INVITE sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKtae51rba. Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9724 INVITE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Supported: replaces User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Type: application/sdp Content-Length: 347 v=0 o=100001102 19218255 19218255 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5006 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5007 <-------------> --- (14 headers 17 lines) --- Sending to 192.168.0.13 : 5060 (no NAT) Using INVITE request as basis request - .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 voip*CLI> <--- Reliably Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKtae51rba.;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as34eaa9ad Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9724 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3c31d375" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '.dw8vu2fm0c72zgb.a3vjf@192.168.0.21' in 32000 ms (Method: INVITE) Found user '100001102' voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ACK sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKtae51rba. Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: ;tag=as34eaa9ad Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9724 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> INVITE sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKe8moz5dnzjj4w Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9725 INVITE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="3c31d375", uri="sip:103@192.168.0.21", response="b1856c02f7485d020ab35bab1c1ea516", algorithm=MD5 Supported: replaces User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Type: application/sdp Content-Length: 347 v=0 o=100001102 19218255 19218255 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5006 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5007 <-------------> --- (15 headers 17 lines) --- Sending to 192.168.0.13 : 5060 (NAT) Using INVITE request as basis request - .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 Found user '100001102' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.13:5006 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.13:5006 Looking for 103 in phones (domain 192.168.0.21) list_route: hop: <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKe8moz5dnzjj4w;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9725 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> -- Executing [103@phones:1] Macro("SIP/100001102-092d6410", "setvars") in new stack -- Executing [s@macro-setvars:1] GotoIf("SIP/100001102-092d6410", "0?blindtransfer") in new stack [Mar 9 17:49:59] DEBUG[5303]: app_macro.c:373 _macro_exec: Executed application: GotoIf -- Executing [s@macro-setvars:2] Set("SIP/100001102-092d6410", "_ACCOUNT=100001") in new stack [Mar 9 17:49:59] DEBUG[5303]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:3] Set("SIP/100001102-092d6410", "SOURCE=102") in new stack [Mar 9 17:49:59] DEBUG[5303]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:4] Set("SIP/100001102-092d6410", "DEST=103") in new stack [Mar 9 17:49:59] DEBUG[5303]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:5] Set("SIP/100001102-092d6410", "CALLERID(num)=102") in new stack [Mar 9 17:49:59] DEBUG[5303]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:6] MacroExit("SIP/100001102-092d6410", "") in new stack -- Executing [103@phones:2] Set("SIP/100001102-092d6410", "CDR(accountcode)=100001") in new stack -- Executing [103@phones:3] Set("SIP/100001102-092d6410", "CDR(userfield)=internal,102 -> 103") in new stack -- Executing [103@phones:4] Macro("SIP/100001102-092d6410", "extension|100001|103") in new stack -- Executing [s@macro-extension:1] Dial("SIP/100001102-092d6410", "SIP/100001103|10") in new stack Audio is at 192.168.0.21 port 19700 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x800 (g726) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.27:20873: INVITE sip:100001103@192.168.0.27:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK4398541b;rport From: "snom" ;tag=as79ddc15f To: Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:49:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 316 v=0 o=root 4964 4964 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19700 RTP/AVP 0 3 8 111 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 100001103 voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK4398541b;received=192.168.0.21;rport=5060 From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 102 INVITE Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/100001103-09305430 is ringing voip*CLI> <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKe8moz5dnzjj4w;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9725 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK4398541b;received=192.168.0.21;rport=5060 From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 102 INVITE Sy.AddrList: 60.241.74.104:20873,192.168.0.27:20873,192.168.254.1:20873,192.168.67.1:20873 Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Type: application/sdp Content-Length: 287 v=0 o=MizuTechSIPS 6367 351 IN IP4 192.168.254.1 s=Mizu c=IN IP4 192.168.254.1 t=0 0 m=audio 30060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=candidates:143983850,192.168.254.1:30060,192.168.0.27:30060,192.168.67.1:30060 <-------------> --- (15 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.1:30060 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x80e (gsm|ulaw|alaw|g726), 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 192.168.254.1:30060 [Mar 9 17:50:01] DEBUG[5116]: chan_sip.c:5692 process_sdp: Oooh, we need to change our audio formats since our peer supports only 0x4 (ulaw) and not 0x2 (gsm) list_route: hop: [Mar 9 17:50:01] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.1, port 20873 Transmitting (NAT) to 192.168.0.27:20873: ACK sip:100001103@192.168.254.1:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK4eb906b7;rport From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/100001103-09305430 answered SIP/100001102-092d6410 Audio is at 192.168.0.21 port 14430 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKe8moz5dnzjj4w;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9725 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Type: application/sdp Content-Length: 262 v=0 o=root 4964 4964 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 14430 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Native bridging SIP/100001102-092d6410 and SIP/100001103-09305430 [Mar 9 17:50:01] DEBUG[5303]: chan_sip.c:6153 reqprep: Strict routing enforced for session 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.1, port 20873 Audio is at 192.168.0.21 port 19700 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.27:20873: INVITE sip:100001103@192.168.254.1:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7345d307;rport From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 237 v=0 o=root 4964 4965 IN IP4 192.168.0.13 s=session c=IN IP4 192.168.0.13 t=0 0 m=audio 5006 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7345d307;received=192.168.0.21;rport=5060 From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 103 INVITE Sy.AddrList: 60.241.74.104:20873,192.168.0.27:20873,192.168.254.1:20873,192.168.67.1:20873 Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Type: application/sdp Content-Length: 287 v=0 o=MizuTechSIPS 6367 352 IN IP4 192.168.254.1 s=Mizu c=IN IP4 192.168.254.1 t=0 0 m=audio 30060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=candidates:143983850,192.168.254.1:30060,192.168.0.27:30060,192.168.67.1:30060 <-------------> --- (15 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.1:30060 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), 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 192.168.254.1:30060 [Mar 9 17:50:01] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.1, port 20873 Transmitting (NAT) to 192.168.0.27:20873: ACK sip:100001103@192.168.254.1:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK3c910321;rport From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ACK sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKkpbc2plzeadhzdpl8a0hjndjak9 Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9725 ACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="3c31d375", uri="sip:103@192.168.0.21", response="60265d131266ae3ce290876e905938b4", algorithm=MD5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Mar 9 17:50:01] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.13, port 5060 Audio is at 192.168.0.21 port 14430 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.13:5060: INVITE sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK0e8db222;rport From: ;tag=as68f3d7e1 To: "snom" ;tag=4bzronvn9d Contact: Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 240 v=0 o=root 4964 4965 IN IP4 192.168.254.1 s=session c=IN IP4 192.168.254.1 t=0 0 m=audio 30060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK0e8db222 From: ;tag=as68f3d7e1 To: "snom" ;tag=4bzronvn9d Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21;rport;branch=z9hG4bK0e8db222 Max-Forwards: 70 From: ;tag=as68f3d7e1 To: "snom" ;tag=4bzronvn9d Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 102 INVITE Contact: Accept: application/sdp Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Type: application/sdp Content-Length: 230 v=0 o=100001102 19218255 19218256 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5006 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5007 <-------------> --- (14 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.13:5006 Found audio description format PCMU for ID 0 Found audio 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 192.168.0.13:5006 [Mar 9 17:50:01] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.13, port 5060 Transmitting (NAT) to 192.168.0.13:5060: ACK sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK72a4553c;rport From: ;tag=as68f3d7e1 To: "snom" ;tag=4bzronvn9d Contact: Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> <-------------> voip*CLI> ******************** PROBLEM HERE No such command '******************** PROBLEM HERE' (type 'help ******************** PROBLEM' for other possible commands) [Mar 9 17:50:07] NOTICE[5116]: chan_sip.c:7685 sip_reregister: -- Re-registration for 61390010834@203.176.185.10 [Mar 9 17:50:07] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:50:07] DEBUG[5116]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '203.176.185.10' AND host = 'dynamic' [Mar 9 17:50:07] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:50:07] DEBUG[5116]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '203.176.185.10' [Mar 9 17:50:07] DEBUG[5116]: chan_sip.c:7922 transmit_register: >>> Re-using Auth data for 61390010834@203.176.185.10 REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 203.176.185.10:5060: REGISTER sip:203.176.185.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7e042d5f;rport From: ;tag=as3c74ddca To: Call-ID: 56c3781d46ae0b8f62547e270ab681c7@192.168.0.73 CSeq: 104 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username="61390010834", realm="203.176.185.10", algorithm=MD5, uri="sip:203.176.185.10", nonce="49b4bc3137d3f9343ed70791af3b127c5d93afcf", response="6a8a4bfdaedb668ab4433cc9acc35264" Expires: 120 Contact: Event: registration Content-Length: 0 --- voip*CLI> <--- SIP read from 203.176.185.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7e042d5f;rport=5060 From: ;tag=as3c74ddca To: ;tag=aca378d4a625ee632e9ca7407d5462e2-e8d0 Call-ID: 56c3781d46ae0b8f62547e270ab681c7@192.168.0.73 CSeq: 104 REGISTER PortaBilling: available-funds:170.33 currency:AUD Contact: ;expires=295 Server: Sip EXpress router (0.9.6 (i386/freebsd)) Content-Length: 0 <-------------> --- (10 headers 0 lines) --- Scheduling destruction of SIP dialog '56c3781d46ae0b8f62547e270ab681c7@192.168.0.73' in 32000 ms (Method: REGISTER) [Mar 9 17:50:07] NOTICE[5116]: chan_sip.c:12890 handle_response_register: Outbound Registration: Expiry for 203.176.185.10 is 295 sec (Scheduling reregistration in 280 s) voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> INVITE sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKnrpes.p174vwef7fvif2cr8px Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9726 INVITE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="3c31d375", uri="sip:103@192.168.0.21", response="b1856c02f7485d020ab35bab1c1ea516", algorithm=MD5 Content-Type: application/sdp Content-Length: 337 v=0 o=100001102 19218255 19218257 IN IP4 0.0.0.0 s=- c=IN IP4 0.0.0.0 t=0 0 a=sendonly m=audio 5006 RTP/AVP 0 8 97 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly a=rtcp:5007 <-------------> --- (13 headers 17 lines) --- Sending to 192.168.0.13 : 5060 (NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 97 Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:5006 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found audio description format iLBC for ID 97 Found audio description format G729 for ID 18 Found audio description format telephone-event for ID 101 Capabilities: us - 0x4 (ulaw), peer - audio=0x50c (ulaw|alaw|g729|ilbc)/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 0.0.0.0:5006 voip*CLI> <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKnrpes.p174vwef7fvif2cr8px;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9726 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> Audio is at 192.168.0.21 port 14430 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKnrpes.p174vwef7fvif2cr8px;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9726 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 4964 4966 IN IP4 192.168.254.1 s=session c=IN IP4 192.168.254.1 t=0 0 m=audio 30060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=recvonly <------------> [Mar 9 17:50:09] DEBUG[5303]: chan_sip.c:6153 reqprep: Strict routing enforced for session 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.1, port 20873 Audio is at 192.168.0.21 port 19700 Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to 192.168.0.27:20873: INVITE sip:100001103@192.168.254.1:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK1437cafc;rport From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 104 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 238 v=0 o=root 4964 4966 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19700 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Started music on hold, class 'default', on SIP/100001103-09305430 voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK1437cafc;received=192.168.0.21;rport=5060 From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 104 INVITE Sy.AddrList: 60.241.74.104:20873,192.168.0.27:20873,192.168.254.1:20873,192.168.67.1:20873 Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Type: application/sdp Content-Length: 287 v=0 o=MizuTechSIPS 6367 353 IN IP4 192.168.254.1 s=Mizu c=IN IP4 192.168.254.1 t=0 0 m=audio 30060 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=sendrecv a=candidates:143983850,192.168.254.1:30060,192.168.0.27:30060,192.168.67.1:30060 <-------------> --- (15 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.254.1:30060 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), 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 192.168.254.1:30060 [Mar 9 17:50:09] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.254.1, port 20873 Transmitting (NAT) to 192.168.0.27:20873: ACK sip:100001103@192.168.254.1:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK0711989d;rport From: "snom" ;tag=as79ddc15f To: ;tag=xe204689002f Contact: Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ACK sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKuqymrh.85xawrvmi9hk3tv8gijj6ty4 Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9726 ACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="3c31d375", uri="sip:103@192.168.0.21", response="60265d131266ae3ce290876e905938b4", algorithm=MD5 ontent-Length: 0 <-------------> --- (9 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> REFER sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKpdj3.lx5wdb9r Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9727 REFER Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="3c31d375", uri="sip:103@192.168.0.21", response="cf0c0f076395c2a24b16927b12182499", algorithm=MD5 Refer-To: Referred-By: Supported: replaces Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Call .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 got a SIP call transfer from caller: (REFER)! SIP transfer to extension 97891942@phones by 100001102@192.168.0.21 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKpdj3.lx5wdb9r;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9727 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 SIP/100001103-09305430 -- Stopped music on hold on SIP/61390010834-092d8820 [Mar 9 17:50:09] DEBUG[5116]: chan_sip.c:13657 attempt_transfer: SIP transfer: Succeeded to masquerade channels. [Mar 9 17:50:09] DEBUG[5116]: chan_sip.c:6153 reqprep: Strict routing enforced for session .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.13, port 5060 Reliably Transmitting (NAT) to 192.168.0.13:5060: NOTIFY sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK1dd24232;rport From: ;tag=as68f3d7e1 To: "snom" ;tag=4bzronvn9d Contact: Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=9727 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 --- [Mar 9 17:50:09] DEBUG[5302]: chan_sip.c:3334 update_call_counter: Call to peer '100001102' removed from call limit 0 Scheduling destruction of SIP dialog '391a41fc3dc6973f2b60ef871beff245@192.168.0.21' in 6400 ms (Method: ACK) [Mar 9 17:50:09] DEBUG[5302]: chan_sip.c:6153 reqprep: Strict routing enforced for session 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.13, port 5060 Reliably Transmitting (NAT) to 192.168.0.13:5060: BYE sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7b93a7bb;rport From: ;tag=as36804318 To: ;tag=xlo563toz4 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Mar 9 17:50:09] DEBUG[5303]: rtp.c:2946 bridge_native_loop: Oooh, something is weird, backing out -- Executing [h@macro-extension:1] GotoIf("SIP/100001102-092d6410", "0?blindtransfer") in new stack -- Executing [h@macro-extension:2] DeadAGI("SIP/100001102-092d6410", "attendedtransfer.agi") in new stack [Mar 9 17:50:09] WARNING[5303]: res_agi.c:2203 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer.agi voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK1dd24232 Max-Forwards: 70 From: ;tag=as68f3d7e1 To: "snom" ;tag=4bzronvn9d Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 103 NOTIFY Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Event: refer;id=9727 Subscription-State: terminated;reason=noresource Supported: replaces User-Agent: Asterisk PBX Content-Length: 0 <-------------> --- (13 headers 0 lines) --- SIP Response message for INCOMING dialog NOTIFY arrived voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK7b93a7bb Max-Forwards: 70 From: ;tag=as36804318 To: ;tag=xlo563toz4 Call-ID: 391a41fc3dc6973f2b60ef871beff245@192.168.0.21 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 <-------------> --- (9 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '391a41fc3dc6973f2b60ef871beff245@192.168.0.21' Method: ACK [Mar 9 17:50:09] DEBUG[5116]: chan_sip.c:3334 update_call_counter: Call to peer '100001102' removed from call limit 0 -- Remote UNIX connection voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> BYE sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKl3anz0eyichyfis Max-Forwards: 70 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9728 BYE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="3c31d375", uri="sip:103@192.168.0.21", response="2f8a67c1ccf78acb3f5f3968a6835aed", algorithm=MD5 Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.0.13 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKl3anz0eyichyfis;received=192.168.0.13 From: "snom" ;tag=4bzronvn9d To: ;tag=as68f3d7e1 Call-ID: .dw8vu2fm0c72zgb.a3vjf@192.168.0.21 CSeq: 9728 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected [Mar 9 17:50:11] ERROR[5303]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe [Mar 9 17:50:11] ERROR[5303]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe -- AGI Script attendedtransfer.agi completed, returning 0 -- Executing [h@macro-extension:3] GotoIf("SIP/100001102-092d6410", "1?attendedtransfer") in new stack -- Goto (macro-extension,h,5) -- Executing [h@macro-extension:5] DeadAGI("SIP/100001102-092d6410", "settransferfield.agi|SIP/100001102-092dddb8|SIP/100001103-09305430") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/settransferfield.agi [Mar 9 17:50:11] ERROR[5303]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe -- AGI Script settransferfield.agi completed, returning 0 -- Executing [h@macro-extension:6] MacroExit("SIP/100001102-092d6410", "") in new stack == Spawn h extension (macro-extension, h, 6) exited non-zero on 'SIP/100001102-092d6410' == Spawn extension (macro-extension, s, 1) exited non-zero on 'SIP/100001102-092d6410' in macro 'extension' == Spawn extension (phones, 103, 4) exited non-zero on 'SIP/100001102-092d6410' [Mar 9 17:50:11] DEBUG[5303]: chan_sip.c:3334 update_call_counter: Call from peer '100001102' removed from call limit 0 Scheduling destruction of SIP dialog '.dw8vu2fm0c72zgb.a3vjf@192.168.0.21' in 32000 ms (Method: BYE) voip*CLI> ******************** PROBLEM END <--- SIP read from 192.168.0.27:20873 ---> REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.27:20873;rport;branch=z9hG4bK-13978701408f-92168025060 From: "sverre" ;tag=192.168.0.21 To: "sverre" Max-Forwards: 70 Call-ID: e1-200093462545c274416453e19216f Sy.Device: C143983850 Sy.LoginName: sverre Sy.NetType: upst CSeq: 13275 REGISTER Contact: Authorization: Digest username="100001103", realm="asterisk", nonce="460e9027", uri="sip:192.168.0.21", response="a00ac4a4a6b31053e695d8d7206b61f9", algorithm=MD5 Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech Accept: application/sdp,application/dtmf-relay,message/sipfrag,text/plain,text/html User-Agent: MizuPhone/1.2.9 FinalUA: MizuPhone Expires: 120 Event: registration Content-Length: 0 <-------------> --- (21 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.27 : 20873 (NAT) <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.27:20873;branch=z9hG4bK-13978701408f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" Call-ID: e1-200093462545c274416453e19216f CSeq: 13275 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.27:20873;branch=z9hG4bK-13978701408f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" ;tag=as34d84d50 Call-ID: e1-200093462545c274416453e19216f CSeq: 13275 REGISTER User-Agent: Asterisk PBX****** PROBLEM END Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4f3c5ece" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e1-200093462545c274416453e19216f' in 32000 ms (Method: REGISTER) voip*CLI> ******************** PROBLEM END <--- SIP read from 192.168.0.27:20873 ---> REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.27:20873;rport;branch=z9hG4bK-56936378380f-92168025060 From: "sverre" ;tag=192.168.0.21 To: "sverre" Max-Forwards: 70 Call-ID: e1-200093462545c274416453e19216f Sy.Device: C143983850 Sy.LoginName: sverre Sy.NetType: upst CSeq: 13276 REGISTER Contact: Authorization: Digest username="100001103", realm="asterisk", nonce="4f3c5ece", uri="sip:192.168.0.21", response="0b8dd27c4f4b2fa1be1779580bd6033b", algorithm=MD5 Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech Accept: application/sdp,application/dtmf-relay,message/sipfrag,text/plain,text/html User-Agent: MizuPhone/1.2.9 FinalUA: MizuPhone Expires: 120 Event: registration Content-Length: 0 <-------------> --- (21 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.27 : 20873 (NAT) <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.27:20873;branch=z9hG4bK-56936378380f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" Call-ID: e1-200093462545c274416453e19216f CSeq: 13276 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Reliably Transmitting (NAT) to 192.168.0.27:20873: OPTIONS sip:100001103@192.168.0.27:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK132c73e9;rport From: "asterisk" ;tag=as3b21192b To: Contact: Call-ID: 3a12d9527644acc770bfd5237c8aec22@192.168.0.21 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:50:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '192.168.0.27', port = '20873', regseconds = '1236581535', username = '100001103', fullcontact = 'sip:100001103@192.168.0.27:20873' WHERE name = '100001103' [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:370 update_mysql: MySQL RealTime: Updated 1 rows on table: sip_buddies [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET lastms = '10' WHERE name = '100001103' [Mar 9 17:50:15] WARNING[5116]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:361 update_mysql: MySQL RealTime: Query: UPDATE sip_buddies SET lastms = '10' WHERE name = '100001103' [Mar 9 17:50:15] DEBUG[5116]: res_config_mysql.c:362 update_mysql: MySQL RealTime: Query Failed because: Unknown column 'lastms' in 'field list' voip*CLI> ******************** PROBLEM END <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.27:20873;branch=z9hG4bK-56936378380f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" ;tag=as34d84d50 Call-ID: e1-200093462545c274416453e19216f CSeq: 13276 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 120 Contact: ;expires=120 Date: Mon, 09 Mar 2009 06:50:15 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e1-200093462545c274416453e19216f' in 32000 ms (Method: REGISTER) voip*CLI> ******************** PROBLEM END <--- SIP read from 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK132c73e9;received=192.168.0.21;rport=5060 From: "asterisk" ;tag=as3b21192b To: ;tag=xe706728036f Contact: Call-ID: 3a12d9527644acc770bfd5237c8aec22@192.168.0.21 CSeq: 102 OPTIONS Sy.AddrList: 60.241.74.104:20873,192.168.0.27:20873,192.168.254.1:20873,192.168.67.1:20873 Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Type: application/sdp Content-Length: 437 v=0 o=MizuTechSIPS 6368 0 IN IP4 192.168.254.1 s=Mizu c=IN IP4 192.168.254.1 t=0 0 m=audio -1 RTP/AVP 106 105 18 4 97 104 0 8 a=rtpmap:106 speex/32000 a=fmtp:106 mode=any a=rtpmap:105 speex/16000 a=fmtp:105 mode=any a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G7231/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=30 a=rtpmap:104 speex/8000 a=fmtp:104 mode=any a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=sendrecv <-------------> --- (15 headers 21 lines) --- Really destroying SIP dialog '3a12d9527644acc770bfd5237c8aec22@192.168.0.21' Method: OPTIONS voip*CLI> ******************** PROBLEM END No such command '******************** PROBLEM END' (type 'help ******************** PROBLEM' for other possible commands) Reliably Transmitting (NAT) to 192.168.0.27:20873: NOTIFY sip:100001103@192.168.254.1:20873 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7062d9ba;rport From: "asterisk" ;tag=as2ec0878d To: ;tag=192.168.0.21 Contact: Call-ID: s5-1352565041788c278901578e1921f CSeq: 105 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 93 Messages-Waiting: yes Message-Account: sip:asterisk@192.168.0.21 Voice-Message: 1/0 (0/0) --- voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK7062d9ba;rport From: "asterisk" ;tag=as2ec0878d To: ;tag=192.168.0.21 Call-ID: s5-1352565041788c278901578e1921f CSeq: 105 NOTIFY Allow: INVITE,REGISTER,UPDATE,OPTIONS,PING,BYE,CANCEL,ACK,COMET,REFER,MESSAGE,SUBSCRIBE,NOTIFY,PUBLISH,INFO,DO,SHAREDFN Allow-Events: presence,refer,telephone-event,keep-alive Supported: privacy,replaces,mizutech User-Agent: MizuPhone/1.2.9 FinalUA: MizuPhone Sy.AddrList: 60.241.74.104:20873,192.168.0.27:20873,192.168.254.1:20873,192.168.67.1:20873 Content-Length: 0 <-------------> --- (13 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ???? <-------------> voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> INFO sip:102@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.1:20873;branch=z9hG4bK-19511469178f-92168025060 From: ;tag=xe204689002f To: "snom" ;tag=as79ddc15f Max-Forwards: 70 Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 184 INFO Sy.Device: S143983850 Sy.LoginName: sverre Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Type: application/dtmf-relay Content-Length: 22 Signal=1 Duration=6 <-------------> --- (13 headers 4 lines) --- Receiving INFO! * DTMF-relay event received: 1 <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.1:20873;branch=z9hG4bK-19511469178f-92168025060;received=192.168.0.27 From: ;tag=xe204689002f To: "snom" ;tag=as79ddc15f Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 184 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- SIP read from 192.168.0.27:20873 ---> INFO sip:102@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.1:20873;branch=z9hG4bK-19511469178f-92168025060 From: ;tag=xe204689002f To: "snom" ;tag=as79ddc15f Max-Forwards: 70 Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 184 INFO Sy.Device: S143983850 Sy.LoginName: sverre Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Type: application/dtmf-relay Content-Length: 22 Signal=3 Duration=6 <-------------> --- (13 headers 4 lines) --- Receiving INFO! <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.1:20873;branch=z9hG4bK-19511469178f-92168025060;received=192.168.0.27 From: ;tag=xe204689002f To: "snom" ;tag=as79ddc15f Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 184 INFO User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> voip*CLI> <--- SIP read from 203.176.185.10:5060 ---> BYE sip:61390010834@192.168.0.21:5060 SIP/2.0 Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKb2d1.71900e82dccd440ca936c6bc4b98b8ff.0 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK1937ca8ea15289ad5892112a9cc808a9;rport=5061 Max-Forwards: 16 From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: ;tag=as476f268a Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 201 BYE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 876175924-925917493-929182820-2 h323-conf-id: 876175924-925917493-929182820-2 H323-credit-time: 7200 <-------------> --- (14 headers 0 lines) --- Sending to 203.176.185.10 : 5060 (no NAT) <--- Transmitting (no NAT) to 203.176.185.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bKb2d1.71900e82dccd440ca936c6bc4b98b8ff.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK1937ca8ea15289ad5892112a9cc808a9;rport=5061 From: ;tag=3c51826fb722e9d8fa4ee14f0e8d2bf4 To: ;tag=as476f268a Call-ID: call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o CSeq: 201 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> -- Executing [h@macro-extension:1] GotoIf("SIP/61390010834-092d8820", "0?blindtransfer") in new stack -- Executing [h@macro-extension:2] DeadAGI("SIP/61390010834-092d8820", "attendedtransfer.agi") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer.agi -- Remote UNIX connection -- Remote UNIX connection disconnected -- Remote UNIX connection voip*CLI> <--- SIP read from 192.168.0.27:20873 ---> BYE sip:102@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.254.1:20873;branch=z9hG4bK-41452060381f-92168025060 From: ;tag=xe204689002f To: "snom" ;tag=as79ddc15f Max-Forwards: 70 Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 264 BYE Reason: SIP ;cause=618 ;text="manual hangup as called" Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Sending to 192.168.0.27 : 20873 (NAT) <--- Transmitting (NAT) to 192.168.0.27:20873 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.254.1:20873;branch=z9hG4bK-41452060381f-92168025060;received=192.168.0.27 From: ;tag=xe204689002f To: "snom" ;tag=as79ddc15f Call-ID: 64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21 CSeq: 264 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> -- Remote UNIX connection disconnected -- Remote UNIX connection -- Remote UNIX connection disconnected -- AGI Script attendedtransfer.agi completed, returning 0 -- Executing [h@macro-extension:3] GotoIf("SIP/61390010834-092d8820", "1?attendedtransfer") in new stack -- Goto (macro-extension,h,5) -- Executing [h@macro-extension:5] DeadAGI("SIP/61390010834-092d8820", "settransferfield.agi|SIP/100001102-092dddb8|SIP/100001103-09305430") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/settransferfield.agi [Mar 9 17:50:21] ERROR[5302]: utils.c:966 ast_carefulwrite: write() returned error: Broken pipe -- AGI Script settransferfield.agi completed, returning 0 -- Executing [h@macro-extension:6] MacroExit("SIP/61390010834-092d8820", "") in new stack == Spawn h extension (macro-extension, h, 6) exited non-zero on 'SIP/61390010834-092d8820' == Spawn extension (macro-extension, s, 1) exited non-zero on 'SIP/61390010834-092d8820' in macro 'extension' == Spawn extension (did, 0390010834, 2) exited non-zero on 'SIP/61390010834-092d8820' [Mar 9 17:50:21] DEBUG[4972]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:50:21] DEBUG[4972]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' AND host = 'dynamic' [Mar 9 17:50:21] DEBUG[4972]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:50:21] DEBUG[4972]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' Really destroying SIP dialog '64b2eec777bb7b297ba910f538ffaeb6@192.168.0.21' Method: BYE Really destroying SIP dialog 'call-F1D8907B-78EE-2B10-0405-37B4F@203.176.186.10~1o' Method: BYE