obelix:~# asterisk -r -vvvv -dddd Parsing /etc/asterisk/asterisk.conf Parsing /etc/asterisk/extconfig.conf == Binding extensions to odbc/asterisk/realtime_extensions == Binding voicemail-extensions to odbc/asterisk/realtime_voicemail_extensions == Binding voicemail to odbc/asterisk/voicemail_users Asterisk CVS-D2005.06.14.22.00.00-07/13/05-15:37:33, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-D2005.06.14.22.00.00-07/13/05-15:37:33 currently running on obelix (pid = 12229) Verbosity is at least 4 -- Remote UNIX connection obelix*CLI> sip debug SIP Debugging enabled obelix*CLI> set verbose 4 Verbosity is at least 4 obelix*CLI> debug level 4 Debugging level set to 4, file '' -- B-channel 0/1 successfully restarted on span 1 -- B-channel 0/2 successfully restarted on span 1 -- B-channel 0/3 successfully restarted on span 1 -- B-channel 0/4 successfully restarted on span 1 -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 -- B-channel 0/7 successfully restarted on span 1 -- B-channel 0/8 successfully restarted on span 1 -- B-channel 0/9 successfully restarted on span 1 -- B-channel 0/10 successfully restarted on span 1 -- B-channel 0/11 successfully restarted on span 1 -- B-channel 0/12 successfully restarted on span 1 -- B-channel 0/13 successfully restarted on span 1 -- B-channel 0/14 successfully restarted on span 1 -- B-channel 0/15 successfully restarted on span 1 -- B-channel 0/17 successfully restarted on span 1 -- B-channel 0/18 successfully restarted on span 1 -- B-channel 0/19 successfully restarted on span 1 -- B-channel 0/20 successfully restarted on span 1 -- B-channel 0/21 successfully restarted on span 1 -- B-channel 0/22 successfully restarted on span 1 -- B-channel 0/23 successfully restarted on span 1 -- B-channel 0/24 successfully restarted on span 1 -- B-channel 0/25 successfully restarted on span 1 -- B-channel 0/26 successfully restarted on span 1 -- B-channel 0/27 successfully restarted on span 1 -- B-channel 0/28 successfully restarted on span 1 -- B-channel 0/29 successfully restarted on span 1 -- B-channel 0/30 successfully restarted on span 1 -- B-channel 0/31 successfully restarted on span 1 obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: INVITE sip:84@voip.sysfrog.org SIP/2.0 Record-Route: Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec From: u6 ;tag=910cf8b324a3aa7co0 To: Call-ID: be6c723f-5b799e60@10.122.32.77 CSeq: 102 INVITE Max-Forwards: 16 Proxy-Authorization: Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@voip.sysfrog.org",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX" Contact: u6 Expires: 240 User-Agent: Sipura/SPA1001-2.0.13(SEg) Content-Length: 426 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 42717460 42717460 IN IP4 XXX.XXX.XXX.XXX s=- c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 32358 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/8000 a=rtpmap:96 G726-40/8000 a=rtpmap:97 G726-24/8000 a=rtpmap:98 G726-16/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (17 headers 19 lines)--- Using INVITE request as basis request - be6c723f-5b799e60@10.122.32.77 Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT) Found peer 'sipsepp' Found RTP audio format 0 Found RTP audio format 2 Found RTP audio format 4 Found RTP audio format 8 Found RTP audio format 18 Found RTP audio format 96 Found RTP audio format 97 Found RTP audio format 98 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port XXX.XXX.XXX.XXX:32358 Found description format PCMU Found description format G726-32 Found description format G723 Found description format PCMA Found description format G729a Found description format G726-40 Found description format G726-24 Found description format G726-16 Found description format NSE Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 84 in incoming-sip list_route: hop: list_route: hop: Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec From: u6 ;tag=910cf8b324a3aa7co0 To: Call-ID: be6c723f-5b799e60@10.122.32.77 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/voip.sysfrog.org-081c3000", "foo=Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@voip.sysfrog.org",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX"") in new stack -- Executing Dial("SIP/voip.sysfrog.org-081c3000", "SIP/u4@voip.sysfrog.org") in new stack We're at ZZZ.ZZZ.ZZZ.ZZZ port 17366 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: INVITE sip:u4@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977 From: "u6" ;tag=as07b5d5de To: Contact: Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Wed, 20 Jul 2005 14:20:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 12249 12249 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=session c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ t=0 0 m=audio 17366 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called u4@voip.sysfrog.org obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977 From: "u6" ;tag=as07b5d5de To: Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 YYY.YYY.YYY.YYY:5060 "Noisy feedback tells: pid=12007 req_src_ip=ZZZ.ZZZ.ZZZ.ZZZ req_src_port=5060 in_uri=sip:u4@voip.sysfrog.org out_uri=sip:u4@XXX.XXX.XXX.XXX:37701 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 180 Ringing From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Record-Route: Content-Length: 0 --- (12 headers 0 lines)--- -- SIP/voip.sysfrog.org-590d is ringing Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec From: u6 ;tag=910cf8b324a3aa7co0 To: ;tag=as43a7adb7 Call-ID: be6c723f-5b799e60@10.122.32.77 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 180 Ringing From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Record-Route: Content-Length: 0 --- (12 headers 0 lines)--- -- SIP/voip.sysfrog.org-590d is ringing obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK74d72977 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Record-Route: Content-Type: application/sdp Content-Length: 218 v=0 o=u4 1121869194 1121869194 IN IP4 XXX.XXX.XXX.XXX s=_ c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 32362 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16 a=ptime:20 a=sendrecv --- (13 headers 11 lines)--- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port XXX.XXX.XXX.XXX:32362 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to YYY.YYY.YYY.YYY, port 5060 Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: ACK sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK6a0b1dcc Route: From: "u6" ;tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Contact: Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-590d answered SIP/voip.sysfrog.org-081c3000 We're at ZZZ.ZZZ.ZZZ.ZZZ port 11478 Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKe335.4af732c3.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-8e60d8ec Record-Route: From: u6 ;tag=910cf8b324a3aa7co0 To: ;tag=as43a7adb7 Call-ID: be6c723f-5b799e60@10.122.32.77 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 12249 12249 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=session c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ t=0 0 m=audio 11478 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/voip.sysfrog.org-081c3000 and SIP/voip.sysfrog.org-590d obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: ACK sip:84@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:1409;branch=z9hG4bK-596204f From: u6 ;tag=910cf8b324a3aa7co0 To: ;tag=as43a7adb7 Call-ID: be6c723f-5b799e60@10.122.32.77 CSeq: 102 ACK Max-Forwards: 16 Proxy-Authorization: Digest username="u6",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:84@ZZZ.ZZZ.ZZZ.ZZZ",algorithm=MD5,response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX" Contact: u6 User-Agent: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (12 headers 0 lines)--- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: INVITE sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 1 INVITE Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKac44.0fb4e8b.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d8d-5c09b94-12fc Max-Forwards: 16 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=u4 1121869194 1121869195 IN IP4 10.122.32.100 s=_ c=IN IP4 0.0.0.0 t=0 0 m=audio 8006 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16 a=ptime:20 a=sendonly --- (14 headers 11 lines)--- Using INVITE request as basis request - 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:8006 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on SIP/voip.sysfrog.org-081c3000 We're at ZZZ.ZZZ.ZZZ.ZZZ port 17366 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKac44.0fb4e8b.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d8d-5c09b94-12fc From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 1 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: ontent-Type: application/sdp Content-Length: 261 v=0 o=root 12249 12250 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=session c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ t=0 0 m=audio 17366 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 20 16:20:04 NOTICE[12249]: res_musiconhold.c:200 ast_moh_files_next: SIP/voip.sysfrog.org-081c3000 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik' obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: ACK sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 1 ACK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d8d-5c09c0c-d85 Max-Forwards: 16 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Content-Length: 0 --- (12 headers 0 lines)--- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: INVITE sip:85@voip.sysfrog.org SIP/2.0 Record-Route: From: "u4";tag=64207a0a-13c442de5d90 To: Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 2 INVITE Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594 Max-Forwards: 16 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Proxy-Authorization: Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5 Content-Type: application/sdp Content-Length: 242 v=0 o=u4 1121869200 1121869201 IN IP4 XXX.XXX.XXX.XXX s=_ c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 32370 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16 a=ptime:20 a=sendrecv --- (16 headers 12 lines)--- Using INVITE request as basis request - 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT) Found peer 'sipsepp' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port XXX.XXX.XXX.XXX:32370 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 85 in incoming-sip list_route: hop: list_route: hop: Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594 From: "u4";tag=64207a0a-13c442de5d90 To: Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Set("SIP/voip.sysfrog.org-081eb528", "foo=Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5") in new stack -- Executing Dial("SIP/voip.sysfrog.org-081eb528", "SIP/u5@voip.sysfrog.org") in new stack We're at ZZZ.ZZZ.ZZZ.ZZZ port 10524 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: INVITE sip:u5@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164 From: "u4" ;tag=as5647a7a8 To: Contact: Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Wed, 20 Jul 2005 14:20:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 12252 12252 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=session c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ t=0 0 m=audio 10524 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called u5@voip.sysfrog.org obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164 From: "u4" ;tag=as5647a7a8 To: Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 YYY.YYY.YYY.YYY:5060 "Noisy feedback tells: pid=12012 req_src_ip=ZZZ.ZZZ.ZZZ.ZZZ req_src_port=5060 in_uri=sip:u5@voip.sysfrog.org out_uri=sip:u5@XXX.XXX.XXX.XXX:1265 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 180 Ringing To: ;tag=37d443a25925fd55i0 From: "u4" ;tag=as5647a7a8 Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164 Record-Route: Server: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/voip.sysfrog.org-fde7 is ringing Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594 From: "u4";tag=64207a0a-13c442de5d90 To: ;tag=as2e10b33f Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK To: ;tag=37d443a25925fd55i0 From: "u4" ;tag=as5647a7a8 Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2b91a164 Record-Route: Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 206 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 657936 657936 IN IP4 XXX.XXX.XXX.XXX s=- c=IN IP4 XXX.XXX.XXX.XXX t=0 0 m=audio 32374 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (12 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port XXX.XXX.XXX.XXX:32374 Found description format PCMU Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to YYY.YYY.YYY.YYY, port 5060 Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: ACK sip:u5@XXX.XXX.XXX.XXX:1265 SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK2a1706c5 Route: From: "u4" ;tag=as5647a7a8 To: ;tag=37d443a25925fd55i0 Contact: Call-ID: 7dddc6614d31222007c83a5211eeb956@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-fde7 answered SIP/voip.sysfrog.org-081eb528 We're at ZZZ.ZZZ.ZZZ.ZZZ port 14242 Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKc9f4.299345b.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d90-5c0a8e6-4594 Record-Route: From: "u4";tag=64207a0a-13c442de5d90 To: ;tag=as2e10b33f Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 12252 12252 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=session c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ t=0 0 m=audio 14242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/voip.sysfrog.org-081eb528 and SIP/voip.sysfrog.org-fde7 obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: ACK sip:85@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: "u4";tag=64207a0a-13c442de5d90 To: ;tag=as2e10b33f Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 2 ACK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d92-5c0afd0-683d Max-Forwards: 16 User-Agent: SIP Phone-1.0.57 Contact: Proxy-Authorization: Digest username="u4", realm="voip.sysfrog.org", nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", uri="sip:85@voip.sysfrog.org", response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX", algorithm=MD5 Content-Length: 0 --- (12 headers 0 lines)--- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: INVITE sip:85@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: "u4";tag=64207a0a-13c442de5d90 To: ;tag=as2e10b33f Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 3 INVITE Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKd9f4.3a305df.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0b9e4-983 Max-Forwards: 16 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Content-Type: application/sdp Content-Length: 237 v=0 o=u4 1121869200 1121869202 IN IP4 10.122.32.100 s=_ c=IN IP4 0.0.0.0 t=0 0 m=audio 8008 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16 a=ptime:20 a=sendonly --- (14 headers 12 lines)--- Using INVITE request as basis request - 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:8008 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) -- Started music on hold, class 'default', on SIP/voip.sysfrog.org-fde7 We're at ZZZ.ZZZ.ZZZ.ZZZ port 14242 Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bKd9f4.3a305df.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0b9e4-983 From: "u4";tag=64207a0a-13c442de5d90 To: ;tag=as2e10b33f Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 214 v=0 o=root 12252 12253 IN IP4 ZZZ.ZZZ.ZZZ.ZZZ s=session c=IN IP4 ZZZ.ZZZ.ZZZ.ZZZ t=0 0 m=audio 14242 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Jul 20 16:20:12 NOTICE[12252]: res_musiconhold.c:200 ast_moh_files_next: SIP/voip.sysfrog.org-fde7 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik' obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: ACK sip:85@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: "u4";tag=64207a0a-13c442de5d90 To: ;tag=as2e10b33f Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 3 ACK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0ba5c-2700 Max-Forwards: 16 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Content-Length: 0 --- (12 headers 0 lines)--- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: REFER sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 2 REFER Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK7c44.12f9be5.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0ba7a-157e Max-Forwards: 16 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Referred-By: Refer-To: Content-Length: 0 --- (15 headers 0 lines)--- Transfer to 85 in incoming-sip-default Transfer from u4 in incoming-sip-default -- Stopped music on hold on SIP/voip.sysfrog.org-081c3000 -- Stopped music on hold on SIP/voip.sysfrog.org-fde7 Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK7c44.12f9be5.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0ba7a-157e From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 2 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to YYY.YYY.YYY.YYY, port 5060 Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: NOTIFY sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK6fd9a84b Route: From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Contact: Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=2 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- set_destination: Parsing for address/port to send to set_destination: set destination to YYY.YYY.YYY.YYY, port 5060 Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: BYE sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK363d538b Route: From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Contact: Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to YYY.YYY.YYY.YYY, port 5060 Reliably Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: BYE sip:u4@XXX.XXX.XXX.XXX:37701 SIP/2.0 Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK5bee0904;rport Route: From: ;tag=as2e10b33f To: "u4";tag=64207a0a-13c442de5d90 Contact: Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 102 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 103 NOTIFY Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK6fd9a84b Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Contact: Content-Length: 0 --- (11 headers 0 lines)--- !!!!!!!---------------************* Why are we here with this packet???? NOTIFY Response message is NOTIFY obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: BYE sip:asterisk@ZZZ.ZZZ.ZZZ.ZZZ SIP/2.0 From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 3 BYE Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK8c44.764533a3.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0bb10-61f5 Max-Forwards: 16 Supported: 100rel,replaces Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE User-Agent: SIP Phone-1.0.57 Content-Length: 0 --- (12 headers 0 lines)--- Sending to YYY.YYY.YYY.YYY : 5060 (non-NAT) Transmitting (no NAT) to YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP YYY.YYY.YYY.YYY;branch=z9hG4bK8c44.764533a3.0 Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:37701;branch=z9hG4bK-42de5d95-5c0bb10-61f5 From: ;tag=64207a0a-13c442de5d8a To: "u6";tag=as07b5d5de Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 3 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK From: "u6";tag=as07b5d5de To: ;tag=64207a0a-13c442de5d8a Call-ID: 07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ CSeq: 104 BYE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;branch=z9hG4bK363d538b Supported: 100rel,replaces User-Agent: SIP Phone-1.0.57 Content-Length: 0 --- (9 headers 0 lines)--- !!!!!!!---------------************* Why are we here with this packet???? BYE Response message is BYE Destroying call '07fe890a761091845295fc2a2934acda@ZZZ.ZZZ.ZZZ.ZZZ' obelix*CLI> <-- SIP read from YYY.YYY.YYY.YYY:5060: SIP/2.0 200 OK From: ;tag=as2e10b33f To: "u4";tag=64207a0a-13c442de5d90 Call-ID: 1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100 CSeq: 102 BYE Via: SIP/2.0/UDP ZZZ.ZZZ.ZZZ.ZZZ:5060;rport=5060;branch=z9hG4bK5bee0904 Supported: 100rel,replaces User-Agent: SIP Phone-1.0.57 Content-Length: 0 --- (9 headers 0 lines)--- !!!!!!!---------------************* Why are we here with this packet???? BYE Response message is BYE Destroying call '1023ce00-64207a0a-13c4-42de5d90-5c0a88c-240b@10.122.32.100' obelix*CLI> quit Executing last minute cleanups Asterisk ending (0). obelix:~# obelix:~#