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=z9hG4bK6951.8546d3e2dcbebbeb7c0f9db155f40588.0 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5824a9314dd6a0d8149a795f04a0b5b8;rport=5061 Max-Forwards: 16 From: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b To: Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o CSeq: 200 INVITE Contact: Anonymous Expires: 300 User-Agent: Sippy cisco-GUID: 876175924-909587761-929118264-2 h323-conf-id: 876175924-909587761-929118264-2 H323-credit-time: 7200 Content-disposition: session Content-Length: 286 Content-Type: application/sdp v=0 o=Sippy 139810860 0 IN IP4 203.176.185.10 s=VoipCall t=0 0 m=audio 57620 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-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o [Mar 9 17:10:04] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5559]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = 'None' 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 203.176.185.10:57620 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 203.176.185.10:57620 Looking for 61390010834 in incoming (domain 192.168.0.21) [Mar 9 17:10:04] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5559]: 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 17:10:04] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5559]: 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=z9hG4bK6951.8546d3e2dcbebbeb7c0f9db155f40588.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5824a9314dd6a0d8149a795f04a0b5b8;rport=5061 Record-Route: From: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b To: Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5532]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5532]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' AND host = 'dynamic' [Mar 9 17:10:04] DEBUG[5532]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5532]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' [Mar 9 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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/61390010834-0a103de8", "did|0390010834|1") in new stack -- Goto (did,0390010834,1) [Mar 9 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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/61390010834-0a103de8", "setgroupifblank|100001") -- Executing [s@macro-setgroupifblank:1] Set("SIP/61390010834-0a103de8", "GROUP()=100001") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:2] NoOp("SIP/61390010834-0a103de8", "") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: NoOp -- Executing [s@macro-setgroupifblank:3] Set("SIP/61390010834-0a103de8", "_ACCOUNT=100001") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:4] Set("SIP/61390010834-0a103de8", "CDR(accountcode)=100001") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:5] Set("SIP/61390010834-0a103de8", "BILLSRC=None") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:6] Set("SIP/61390010834-0a103de8", "BILLDST=0390010834") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:7] Set("SIP/61390010834-0a103de8", "CDR(userfield)=incoming,None -> 0390010834") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setgroupifblank:8] NoOp("SIP/61390010834-0a103de8", "") in new stack [Mar 9 17:10:04] DEBUG[5602]: app_macro.c:373 _macro_exec: Executed application: NoOp [Mar 9 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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/61390010834-0a103de8", "extension|100001|102") -- Executing [s@macro-extension:1] Dial("SIP/61390010834-0a103de8", "SIP/100001102|10") in new stack Audio is at 192.168.0.21 port 15204 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=z9hG4bK2ee3bd24;rport From: "anonymous" ;tag=as27ef84d1 To: Contact: Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:10:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 316 v=0 o=root 5528 5528 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 15204 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 17:10:04] DEBUG[5602]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:04] DEBUG[5602]: 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 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=z9hG4bK2ee3bd24 From: "anonymous" ;tag=as27ef84d1 To: Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@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/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK2ee3bd24 Max-Forwards: 70 From: "anonymous" ;tag=as27ef84d1 To: ;tag=0ulq.wt8syr8 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 102 INVITE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- SIP/100001102-0a0e9320 is ringing <--- Transmitting (no NAT) to 203.176.185.10:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 203.176.185.10;branch=z9hG4bK6951.8546d3e2dcbebbeb7c0f9db155f40588.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5824a9314dd6a0d8149a795f04a0b5b8;rport=5061 Record-Route: From: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b To: ;tag=as05a85f64 Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@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=z9hG4bK2ee3bd24 Max-Forwards: 70 From: "anonymous" ;tag=as27ef84d1 To: ;tag=0ulq.wt8syr8 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@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 118521583 118521583 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5018 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:5019 <-------------> --- (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 192.168.0.13:5018 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 192.168.0.13:5018 [Mar 9 17:10:06] DEBUG[5559]: 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 17:10:06] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session 0c0bfcc901bc0b3b4cf830d85e4ca964@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=z9hG4bK7bcb87f4;rport From: "anonymous" ;tag=as27ef84d1 To: ;tag=0ulq.wt8syr8 Contact: Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/100001102-0a0e9320 answered SIP/61390010834-0a103de8 Audio is at 192.168.0.21 port 19932 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=z9hG4bK6951.8546d3e2dcbebbeb7c0f9db155f40588.0;received=203.176.185.10 Via: SIP/2.0/UDP 203.176.185.10:5061;branch=z9hG4bK5824a9314dd6a0d8149a795f04a0b5b8;rport=5061 Record-Route: From: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b To: ;tag=as05a85f64 Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@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 5528 5528 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19932 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/61390010834-0a103de8 and SIP/100001102-0a0e9320 [Mar 9 17:10:06] DEBUG[5532]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:06] DEBUG[5532]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' AND host = 'dynamic' [Mar 9 17:10:06] DEBUG[5532]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:06] DEBUG[5532]: 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=z9hG4bK6951.f00858e1162dc3cad263c11d12913d71.0 Via: SIP/2.0/UDP 203.176.185.10:5061;rport=5061;branch=z9hG4bK6247b19f9ba68797d7d7dbf79c86fba5 Max-Forwards: 16 From: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b To: ;tag=as05a85f64 Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o CSeq: 200 ACK Expires: 300 User-Agent: Sippy <-------------> --- (10 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> INVITE sip:None@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK1y973cl74zv.q7xvv5mqo9or57a Max-Forwards: 70 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1115 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 118521583 118521584 IN IP4 0.0.0.0 s=- c=IN IP4 0.0.0.0 t=0 0 a=sendonly m=audio 5018 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:5019 <-------------> --- (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 0.0.0.0:5018 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 0.0.0.0:5018 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK1y973cl74zv.q7xvv5mqo9or57a;received=192.168.0.13 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1115 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 15204 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=z9hG4bK1y973cl74zv.q7xvv5mqo9or57a;received=192.168.0.13 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1115 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 5528 5529 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 15204 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/61390010834-0a103de8 [Mar 9 17:10:08] DEBUG[5602]: rtp.c:3222 bridge_p2p_loop: Oooh, formats changed, backing out voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ACK sip:None@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKdetzw4nvs4b.wbu Max-Forwards: 70 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1115 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=z9hG4bKrchhb6dkao1 Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3009 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: 349 v=0 o=100001102 545559021 545559021 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5020 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:5021 <-------------> --- (14 headers 17 lines) --- Sending to 192.168.0.13 : 5060 (no NAT) Using INVITE request as basis request - e6mz7ep3k5z6lgb433x4@192.168.0.21 <--- 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=z9hG4bKrchhb6dkao1;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as7802409a Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3009 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0c0a3237" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e6mz7ep3k5z6lgb433x4@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=z9hG4bKrchhb6dkao1 Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as7802409a Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3009 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=z9hG4bKfjxm3w5hqm3oevs1v7dv6o1 Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3010 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="0c0a3237", uri="sip:103@192.168.0.21", response="0a467b9a1d88ae2f2009f59c0a9be8ae", algorithm=MD5 Supported: replaces User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Type: application/sdp Content-Length: 349 v=0 o=100001102 545559021 545559021 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5020 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:5021 <-------------> --- (15 headers 17 lines) --- Sending to 192.168.0.13 : 5060 (NAT) Using INVITE request as basis request - e6mz7ep3k5z6lgb433x4@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 192.168.0.13:5020 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 192.168.0.13:5020 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=z9hG4bKfjxm3w5hqm3oevs1v7dv6o1;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3010 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/100001102-0a1210f8", "setvars") in new stack -- Executing [s@macro-setvars:1] GotoIf("SIP/100001102-0a1210f8", "0?blindtransfer") in new stack [Mar 9 17:10:10] DEBUG[5604]: app_macro.c:373 _macro_exec: Executed application: GotoIf -- Executing [s@macro-setvars:2] Set("SIP/100001102-0a1210f8", "_ACCOUNT=100001") in new stack [Mar 9 17:10:10] DEBUG[5604]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:3] Set("SIP/100001102-0a1210f8", "SOURCE=102") in new stack [Mar 9 17:10:10] DEBUG[5604]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:4] Set("SIP/100001102-0a1210f8", "DEST=103") in new stack [Mar 9 17:10:10] DEBUG[5604]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:5] Set("SIP/100001102-0a1210f8", "CALLERID(num)=102") in new stack [Mar 9 17:10:10] DEBUG[5604]: app_macro.c:373 _macro_exec: Executed application: Set -- Executing [s@macro-setvars:6] MacroExit("SIP/100001102-0a1210f8", "") in new stack -- Executing [103@phones:2] Set("SIP/100001102-0a1210f8", "CDR(accountcode)=100001") in new stack -- Executing [103@phones:3] Set("SIP/100001102-0a1210f8", "CDR(userfield)=internal,102 -> 103") in new stack -- Executing [103@phones:4] Macro("SIP/100001102-0a1210f8", "extension|100001|103") in new stack -- Executing [s@macro-extension:1] Dial("SIP/100001102-0a1210f8", "SIP/100001103|10") in new stack Audio is at 192.168.0.21 port 10198 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=z9hG4bK4e53b3de;rport From: "snom" ;tag=as749fc5b0 To: Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:10:10 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 316 v=0 o=root 5528 5528 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 10198 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=z9hG4bK4e53b3de;received=192.168.0.21;rport=5060 From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 102 INVITE Server: MizuPhone/1.2.9 FinalUA: MizuPhone Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- SIP/100001103-0a0f3dd0 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=z9hG4bKfjxm3w5hqm3oevs1v7dv6o1;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3010 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=z9hG4bK4e53b3de;received=192.168.0.21;rport=5060 From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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 6304 288 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 17:10:11] DEBUG[5559]: 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 17:10:11] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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=z9hG4bK7155aede;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- SIP/100001103-0a0f3dd0 answered SIP/100001102-0a1210f8 Audio is at 192.168.0.21 port 19624 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=z9hG4bKfjxm3w5hqm3oevs1v7dv6o1;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3010 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 5528 5528 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19624 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/100001102-0a1210f8 and SIP/100001103-0a0f3dd0 [Mar 9 17:10:11] DEBUG[5604]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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 10198 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=z9hG4bK78edffe1;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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 5528 5529 IN IP4 192.168.0.13 s=session c=IN IP4 192.168.0.13 t=0 0 m=audio 5020 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=z9hG4bK78edffe1;received=192.168.0.21;rport=5060 From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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 6304 289 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 17:10:11] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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=z9hG4bK0b6d0e7a;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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=z9hG4bK9hefr3f07xcannfyppavhhuzi Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3010 ACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="0c0a3237", uri="sip:103@192.168.0.21", response="eb069a75ddd67026b26607be87f25a2e", algorithm=MD5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- [Mar 9 17:10:11] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session e6mz7ep3k5z6lgb433x4@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 19624 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=z9hG4bK5eb0511d;rport From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Contact: Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 Seq: 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 5528 5529 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=z9hG4bK5eb0511d From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@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=z9hG4bK5eb0511d Max-Forwards: 70 From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@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: 232 v=0 o=100001102 545559021 545559022 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5020 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5021 <-------------> --- (14 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.13:5020 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 192.168.0.13:5020 [Mar 9 17:10:11] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session e6mz7ep3k5z6lgb433x4@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=z9hG4bK600b3f15;rport From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Contact: Call-ID: e6mz7ep3k5z6lgb433x4@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 ---> REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.27:20873;rport;branch=z9hG4bK-17738886334f-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: 13185 REGISTER Contact: Authorization: Digest username="100001103", realm="asterisk", nonce="52db86f9", uri="sip:192.168.0.21", response="94d692b98821d94cddbba24012de8a0f", 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-17738886334f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" Call-ID: e1-200093462545c274416453e19216f CSeq: 13185 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-17738886334f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" ;tag=as74231b0e Call-ID: e1-200093462545c274416453e19216f CSeq: 13185 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="30047831" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e1-200093462545c274416453e19216f' in 32000 ms (Method: REGISTER) voip*CLI> <--- 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-46374061743f-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: 13186 REGISTER Contact: Authorization: Digest username="100001103", realm="asterisk", nonce="30047831", uri="sip:192.168.0.21", response="b95725e17b62025f06a9696a316f49a3", 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-46374061743f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" Call-ID: e1-200093462545c274416453e19216f CSeq: 13186 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=z9hG4bK6211ae3e;rport From: "asterisk" ;tag=as5e3c8773 To: Contact: Call-ID: 16aa59d43dc304246754e9ee56885dbc@192.168.0.21 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:10:11 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '192.168.0.27', port = '20873', regseconds = '1236579131', username = '100001103', fullcontact = 'sip:100001103@192.168.0.27:20873' WHERE name = '100001103' [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:370 update_mysql: MySQL RealTime: Updated 1 rows on table: sip_buddies [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET lastms = '10' WHERE name = '100001103' [Mar 9 17:10:11] WARNING[5559]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:361 update_mysql: MySQL RealTime: Query: UPDATE sip_buddies SET lastms = '10' WHERE name = '100001103' [Mar 9 17:10:11] DEBUG[5559]: res_config_mysql.c:362 update_mysql: MySQL RealTime: Query Failed because: Unknown column 'lastms' in 'field list' <--- 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-46374061743f-92168025060;received=192.168.0.27;rport=20873 From: "sverre" ;tag=192.168.0.21 To: "sverre" ;tag=as74231b0e Call-ID: e1-200093462545c274416453e19216f Seq: 13186 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 06:10:11 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'e1-200093462545c274416453e19216f' in 32000 ms (Method: REGISTER) 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=z9hG4bK6211ae3e;received=192.168.0.21;rport=5060 From: "asterisk" ;tag=as5e3c8773 To: ;tag=xe101372422f Contact: Call-ID: 16aa59d43dc304246754e9ee56885dbc@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 6305 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 '16aa59d43dc304246754e9ee56885dbc@192.168.0.21' Method: OPTIONS 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=z9hG4bK51192c52;rport From: "asterisk" ;tag=as39d9ca5e To: ;tag=192.168.0.21 Contact: Call-ID: s5-1278643928418c276554921e1921f CSeq: 104 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=z9hG4bK51192c52;rport From: "asterisk" ;tag=as39d9ca5e To: ;tag=192.168.0.21 Call-ID: s5-1278643928418c276554921e1921f CSeq: 104 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> ******************** PROBLEM HERE No such command '******************** PROBLEM HERE' (type 'help ******************** PROBLEM' for other possible commands) 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=z9hG4bKp7xk2klb.4gym.fskc1 Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3011 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="0c0a3237", uri="sip:103@192.168.0.21", response="0a467b9a1d88ae2f2009f59c0a9be8ae", algorithm=MD5 Content-Type: application/sdp Content-Length: 339 v=0 o=100001102 545559021 545559023 IN IP4 0.0.0.0 s=- c=IN IP4 0.0.0.0 t=0 0 a=sendonly m=audio 5020 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:5021 <-------------> --- (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 0.0.0.0:5020 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 0.0.0.0:5020 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKp7xk2klb.4gym.fskc1;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3011 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 19624 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=z9hG4bKp7xk2klb.4gym.fskc1;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3011 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 5528 5530 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 17:10:16] DEBUG[5604]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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 10198 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=z9hG4bK2f6c828b;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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 5528 5530 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 10198 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/100001103-0a0f3dd0 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=z9hG4bK2f6c828b;received=192.168.0.21;rport=5060 From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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 6304 290 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 17:10:16] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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=z9hG4bK27403976;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@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=z9hG4bK3216fol9dxi4jji Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3011 ACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="0c0a3237", uri="sip:103@192.168.0.21", response="eb069a75ddd67026b26607be87f25a2e", algorithm=MD5 Content-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=z9hG4bKbac2bmtd6m2 Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3012 REFER Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="0c0a3237", uri="sip:103@192.168.0.21", response="d1190a2654eaede9311386967860265a", algorithm=MD5 Refer-To: Referred-By: Supported: replaces Content-Length: 0 <-------------> --- (14 headers 0 lines) --- Call e6mz7ep3k5z6lgb433x4@192.168.0.21 got a SIP call transfer from caller: (REFER)! Failed SIP Transfer to non-existing extension None in context phones n <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKbac2bmtd6m2;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3012 REFER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Contact: Content-Length: 0 <------------> [Mar 9 17:10:16] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session e6mz7ep3k5z6lgb433x4@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=z9hG4bK6d5f8a67;rport From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Contact: Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 103 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: refer;id=3012 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: 23 SIP/2.0 404 Not found --- 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=z9hG4bK6d5f8a67 Max-Forwards: 70 From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 103 NOTIFY Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Event: refer;id=3012 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 ---> INVITE sip:103@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK38bk1ie3su4 Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3013 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="0c0a3237", uri="sip:103@192.168.0.21", response="0a467b9a1d88ae2f2009f59c0a9be8ae", algorithm=MD5 Content-Type: application/sdp Content-Length: 339 v=0 o=100001102 545559021 545559024 IN IP4 0.0.0.0 s=- c=IN IP4 0.0.0.0 t=0 0 a=sendonly m=audio 5020 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:5021 <-------------> --- (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 0.0.0.0:5020 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 0.0.0.0:5020 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK38bk1ie3su4;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3013 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 19624 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=z9hG4bK38bk1ie3su4;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3013 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 5528 5531 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 <------------> -- Stopped music on hold on SIP/100001103-0a0f3dd0 -- Started music on hold, class 'default', on SIP/100001103-0a0f3dd0 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=z9hG4bKcwq1ohu962cab9zgl2scil6jl Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3013 ACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="0c0a3237", uri="sip:103@192.168.0.21", response="eb069a75ddd67026b26607be87f25a2e", algorithm=MD5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> INVITE sip:None@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKos5cickj.2p3v Max-Forwards: 70 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1116 INVITE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Disposition: session Content-Type: application/sdp Content-Length: 349 v=0 o=100001102 118521583 118521585 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5018 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:5019 <-------------> --- (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 192.168.0.13:5018 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 192.168.0.13:5018 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKos5cickj.2p3v;received=192.168.0.13 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1116 INVITE ser-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 15204 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=z9hG4bKos5cickj.2p3v;received=192.168.0.13 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1116 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 5528 5530 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 15204 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 <------------> -- Stopped music on hold on SIP/61390010834-0a103de8 voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> ACK sip:None@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK64u.o76ok4i5opdqcmx5wit1b9u56ku Max-Forwards: 70 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1116 ACK Content-Length: 0 <-------------> --- (8 headers 0 lines) --- voip*CLI> ******************** START HERE <--- SIP read from 192.168.0.13:5060 ---> REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKjn9a60rcb7vsica547o80dfrz.iww Max-Forwards: 70 From: ;tag=y.mck. To: Call-ID: 6tbz12r9rqlst66.ve.jazdt0hk0 CSeq: 182453 REGISTER Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Expires: 600 User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.13 : 5060 (no NAT) <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKjn9a60rcb7vsica547o80dfrz.iww;received=192.168.0.13 From: ;tag=y.mck. To: Call-ID: 6tbz12r9rqlst66.ve.jazdt0hk0 CSeq: 182453 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKjn9a60rcb7vsica547o80dfrz.iww;received=192.168.0.13 From: ;tag=y.mck. To: ;tag=as234c84dc Call-ID: 6tbz12r9rqlst66.ve.jazdt0hk0 CSeq: 182453 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="0e2c17b4" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6tbz12r9rqlst66.ve.jazdt0hk0' in 32000 ms (Method: REGISTER) voip*CLI> ******************** START HERE <--- SIP read from 192.168.0.13:5060 ---> REGISTER sip:192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK0iezwyrndxybbz.o4 Max-Forwards: 70 From: ;tag=y.mck. To: Call-ID: 6tbz12r9rqlst66.ve.jazdt0hk0 CSeq: 182454 REGISTER Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Authorization: Digest username="100001102", realm="asterisk", nonce="0e2c17b4", uri="sip:192.168.0.21", response="2ad55c0268d89181ca3108c8dd6d669a", algorithm=MD5 Expires: 600 User-Agent: snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1) Content-Length: 0 <-------------> --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.0.13 : 5060 (NAT) <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK0iezwyrndxybbz.o4;received=192.168.0.13 From: ;tag=y.mck. To: Call-ID: 6tbz12r9rqlst66.ve.jazdt0hk0 CSeq: 182454 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> Reliably Transmitting (NAT) to 192.168.0.13:5060: OPTIONS sip:100001102@192.168.0.13 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK6dd3b2d5;rport From: "asterisk" ;tag=as27093c56 To: Contact: Call-ID: 0d2087c20a7e74843fafb65646d66851@192.168.0.21 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Mon, 09 Mar 2009 06:10:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- -- Saved useragent "snom-m3-SIP/01.22 (MAC=0004132A01BD; HW=1)" for peer 100001102 [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '192.168.0.13', port = '5060', regseconds = '1236579618', username = '100001102', fullcontact = 'sip:100001102@192.168.0.13' WHERE name = '100001102' [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:370 update_mysql: MySQL RealTime: Updated 1 rows on table: sip_buddies [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET lastms = '0' WHERE name = '100001102' [Mar 9 17:10:18] WARNING[5559]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:361 update_mysql: MySQL RealTime: Query: UPDATE sip_buddies SET lastms = '0' WHERE name = '100001102' [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:362 update_mysql: MySQL RealTime: Query Failed because: Unknown column 'lastms' in 'field list' <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bK0iezwyrndxybbz.o4;received=192.168.0.13 From: ;tag=y.mck. To: ;tag=as234c84dc Call-ID: 6tbz12r9rqlst66.ve.jazdt0hk0 CSeq: 182454 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 600 Contact: ;expires=600 Date: Mon, 09 Mar 2009 06:10:18 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog '6tbz12r9rqlst66.ve.jazdt0hk0' in 32000 ms (Method: REGISTER) voip*CLI> ******************** START HERE <--- 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=z9hG4bK6dd3b2d5 Max-Forwards: 70 From: "asterisk" ;tag=as27093c56 To: Call-ID: 0d2087c20a7e74843fafb65646d66851@192.168.0.21 CSeq: 102 OPTIONS Contact: Accept: application/sdp Accept-Encoding: identity Accept-Language: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Date: Mon, 09 Mar 2009 06:10:18 GMT Supported: replaces User-Agent: Asterisk PBX Content-Length: 0 <-------------> --- (16 headers 0 lines) --- [Mar 9 17:10:18] NOTICE[5559]: chan_sip.c:12925 handle_response_peerpoke: Peer '100001102' is now Reachable. (10ms / 2000ms) [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:356 update_mysql: MySQL RealTime: Update SQL: UPDATE sip_buddies SET lastms = '10' WHERE name = '100001102' [Mar 9 17:10:18] WARNING[5559]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:361 update_mysql: MySQL RealTime: Query: UPDATE sip_buddies SET lastms = '10' WHERE name = '100001102' [Mar 9 17:10:18] DEBUG[5559]: res_config_mysql.c:362 update_mysql: MySQL RealTime: Query Failed because: Unknown column 'lastms' in 'field list' Really destroying SIP dialog '0d2087c20a7e74843fafb65646d66851@192.168.0.21' Method: OPTIONS voip*CLI> ******************** START HERE <--- SIP read from 192.168.0.27:20873 ---> <-------------> voip*CLI> ******************** PROBLEM END No such command '******************** PROBLEM END' (type 'help ******************** PROBLEM' for other possible commands) voip*CLI> <--- SIP read from 192.168.0.13:5060 ---> BYE sip:None@192.168.0.21 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKbsglnxe0l.ei.pnn88rrbty Max-Forwards: 70 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1117 BYE Contact: Allow: INVITE, CANCEL, BYE, ACK, REGISTER, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE, INFO, PRACK Content-Length: 0 <-------------> --- (10 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=z9hG4bKbsglnxe0l.ei.pnn88rrbty;received=192.168.0.13 From: ;tag=0ulq.wt8syr8 To: ;tag=as27ef84d1 Call-ID: 0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21 CSeq: 1117 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/61390010834-0a103de8", "0?blindtransfer") in new stack -- Executing [h@macro-extension:2] DeadAGI("SIP/61390010834-0a103de8", "attendedtransfer.agi") in new stack [Mar 9 17:10:21] WARNING[5602]: 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 -- Remote UNIX connection 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=z9hG4bKyam4j7pwrwfa68dyvuyg8igt. Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3014 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="0c0a3237", uri="sip:103@192.168.0.21", response="0a467b9a1d88ae2f2009f59c0a9be8ae", algorithm=MD5 Content-Type: application/sdp Content-Length: 349 v=0 o=100001102 545559021 545559025 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5020 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:5021 <-------------> --- (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 192.168.0.13:5020 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 192.168.0.13:5020 <--- Transmitting (NAT) to 192.168.0.13:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.13;branch=z9hG4bKyam4j7pwrwfa68dyvuyg8igt.;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3014 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 19624 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=z9hG4bKyam4j7pwrwfa68dyvuyg8igt.;received=192.168.0.13 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3014 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 5528 5532 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 <------------> [Mar 9 17:10:22] DEBUG[5604]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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 10198 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=z9hG4bK4cf96077;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 105 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 5528 5531 IN IP4 192.168.0.13 s=session c=IN IP4 192.168.0.13 t=0 0 m=audio 5020 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 --- -- Stopped music on hold on SIP/100001103-0a0f3dd0 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=z9hG4bK4cf96077;received=192.168.0.21;rport=5060 From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 105 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 6304 291 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 - 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 192.168.254.1:30060 [Mar 9 17:10:22] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session 06cf604818b7ffe67d5decc2353141d2@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=z9hG4bK05527a6f;rport From: "snom" ;tag=as749fc5b0 To: ;tag=xe402250873f Contact: Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 105 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=z9hG4bK4.zywjztc Max-Forwards: 70 From: "snom" ;tag=rwxfbx To: ;tag=as3e59f2cc Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 3014 ACK Proxy-Authorization: Digest username="100001102", realm="asterisk", nonce="0c0a3237", uri="sip:103@192.168.0.21", response="eb069a75ddd67026b26607be87f25a2e", algorithm=MD5 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- -- 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/61390010834-0a103de8", "0?attendedtransfer") in new stack -- Executing [h@macro-extension:4] MacroExit("SIP/61390010834-0a103de8", "") in new stack == Spawn h extension (macro-extension, h, 4) exited non-zero on 'SIP/61390010834-0a103de8' == Spawn extension (macro-extension, s, 1) exited non-zero on 'SIP/61390010834-0a103de8' in macro 'extension' == Spawn extension (did, 0390010834, 2) exited non-zero on 'SIP/61390010834-0a103de8' Scheduling destruction of SIP dialog 'call-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o' in 6400 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 203.176.185.10, port 5060 Reliably Transmitting (no NAT) to 203.176.185.10:5060: BYE sip:203.176.185.10:5061 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK28b44535;rport Route: From: ;tag=as05a85f64 To: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o CSeq: 102 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Mar 9 17:10:22] DEBUG[5532]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:22] DEBUG[5532]: res_config_mysql.c:140 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '61390010834' AND host = 'dynamic' [Mar 9 17:10:22] DEBUG[5532]: res_config_mysql.c:653 mysql_reconnect: MySQL RealTime: Everything is fine. [Mar 9 17:10:22] DEBUG[5532]: 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 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.21:5060;branch=z9hG4bK28b44535;rport=5060 From: ;tag=as05a85f64 To: anonymous ;tag=7c2dc3edeb530f73cd3bd6ae102a3d6b Call-ID: call-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o CSeq: 102 BYE Server: Sippy <-------------> --- (7 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '0c0bfcc901bc0b3b4cf830d85e4ca964@192.168.0.21' Method: BYE Really destroying SIP dialog 'call-F1D6A460-6DEE-2B10-030F-37A8A@203.176.186.10~1o' Method: ACK 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-17116018664f-92168025060 From: ;tag=xe402250873f To: "snom" ;tag=as749fc5b0 Max-Forwards: 70 Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 185 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-17116018664f-92168025060;received=192.168.0.27 From: ;tag=xe402250873f To: "snom" ;tag=as749fc5b0 Call-ID: 06cf604818b7ffe67d5decc2353141d2@192.168.0.21 CSeq: 185 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 <------------> [Mar 9 17:10:24] DEBUG[5604]: chan_sip.c:6153 reqprep: Strict routing enforced for session e6mz7ep3k5z6lgb433x4@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 19624 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=z9hG4bK0760f0d1;rport From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Contact: Call-ID: e6mz7ep3k5z6lgb433x4@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 5528 5533 IN IP4 192.168.0.21 s=session c=IN IP4 192.168.0.21 t=0 0 m=audio 19624 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 --- -- Executing [h@macro-extension:1] GotoIf("SIP/100001102-0a1210f8", "0?blindtransfer") in new stack -- Executing [h@macro-extension:2] DeadAGI("SIP/100001102-0a1210f8", "attendedtransfer.agi") in new stack [Mar 9 17:10:24] WARNING[5604]: 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/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.21;rport=5060;branch=z9hG4bK0760f0d1 From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 104 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=z9hG4bK0760f0d1 Max-Forwards: 70 From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 104 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: 232 v=0 o=100001102 545559021 545559026 IN IP4 192.168.0.13 s=- c=IN IP4 192.168.0.13 t=0 0 a=sendrecv m=audio 5020 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=rtcp:5021 <-------------> --- (14 headers 12 lines) --- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.13:5020 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 192.168.0.13:5020 [Mar 9 17:10:24] DEBUG[5559]: chan_sip.c:6153 reqprep: Strict routing enforced for session e6mz7ep3k5z6lgb433x4@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=z9hG4bK0b9889e9;rport From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Contact: Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 104 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Remote UNIX connection -- 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/100001102-0a1210f8", "0?attendedtransfer") in new stack -- Executing [h@macro-extension:4] MacroExit("SIP/100001102-0a1210f8", "") in new stack == Spawn h extension (macro-extension, h, 4) exited non-zero on 'SIP/100001102-0a1210f8' == Spawn extension (macro-extension, s, 1) exited non-zero on 'SIP/100001102-0a1210f8' in macro 'extension' == Spawn extension (phones, 103, 4) exited non-zero on 'SIP/100001102-0a1210f8' Scheduling destruction of SIP dialog 'e6mz7ep3k5z6lgb433x4@192.168.0.21' in 32000 ms (Method: ACK) [Mar 9 17:10:25] DEBUG[5604]: chan_sip.c:6153 reqprep: Strict routing enforced for session e6mz7ep3k5z6lgb433x4@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=z9hG4bK0a831dae;rport From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 105 BYE User-Agent: Asterisk PBX Max-Forwards: 70 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 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=5060;branch=z9hG4bK0a831dae Max-Forwards: 70 From: ;tag=as3e59f2cc To: "snom" ;tag=rwxfbx Call-ID: e6mz7ep3k5z6lgb433x4@192.168.0.21 CSeq: 105 BYE User-Agent: Asterisk PBX Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Really destroying SIP dialog '06cf604818b7ffe67d5decc2353141d2@192.168.0.21' Method: BYE Really destroying SIP dialog 'e6mz7ep3k5z6lgb433x4@192.168.0.21' Method: ACK Scheduling destruction of SIP dialog '5a34fb2170df54f52ea2591915f74307@192.168.0.21' in 6400 ms (Method: NOTIFY) 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=z9hG4bK31200c78;rport From: "asterisk" ;tag=as10ee57b5 To: Contact: Call-ID: 5a34fb2170df54f52ea2591915f74307@192.168.0.21 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 92 Messages-Waiting: no Message-Account: sip:asterisk@192.168.0.21 Voice-Message: 0/0 (0/0) --- 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=5060;branch=z9hG4bK31200c78 Max-Forwards: 70 From: "asterisk" ;tag=as10ee57b5 To: Call-ID: 5a34fb2170df54f52ea2591915f74307@192.168.0.21 CSeq: 102 NOTIFY Contact: Event: message-summary User-Agent: Asterisk PBX Content-Length: 0 <-------------> --- (11 headers 0 lines) --- Really destroying SIP dialog '5a34fb2170df54f52ea2591915f74307@192.168.0.21' Method: NOTIFY