samuel*CLI> set verbose 4 Verbosity was 0 and is now 4 samuel*CLI> set debug 4 Core debug was 0 and is now 4 samuel*CLI> samuel*CLI> samuel*CLI> samuel*CLI> samuel*CLI> samuel*CLI> sip debug SIP Debugging enabled samuel*CLI> <-- SIP read from 111.222.32.83:5060: INVITE sip:371@asterisk-cvs.at SIP/2.0 Record-Route: Via: SIP/2.0/UDP 111.222.32.83;branch=z9hG4bK724e.58ff7932.0 Via: SIP/2.0/UDP 111.222.32.80:52613;received=111.222.32.83;branch=z9hG4bK.0fa56909;rport=52614 Max-Forwards: 15 Contact: To: ;tag=12345 From: "Klaus Darilion enum eyebea";tag=f1614469 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: sipsak Content-Length: 210 P-Behind-NAT: Yes v=0 o=- 23084192 23084377 IN IP4 10.10.0.50 s=eyeBeam c=IN IP4 111.222.32.83 t=0 0 m=audio 35062 RTP/AVP 0 8 3 98 101 a=fmtp:101 0-15 a=rtpmap:98 ilbc/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv --- (15 headers 10 lines)--- Using INVITE request as basis request - 03614d34b41caf24@a2xhdXNwYw.. Sending to 111.222.32.83 : 5060 (non-NAT) Found no matching peer or user for '111.222.32.83:5060' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 98 Found RTP audio format 101 Peer audio RTP is at port 111.222.32.83:35062 Found description format ilbc Found description format telephone-event Capabilities: us - 0x8060e (gsm|ulaw|alaw|speex|ilbc|h263), peer - audio=0x40e (gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0x40e (gsm|ulaw|alaw|ilbc) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 371 in sipanon list_route: hop: list_route: hop: Transmitting (no NAT) to 111.222.32.83:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.222.32.83;branch=z9hG4bK724e.58ff7932.0 Via: SIP/2.0/UDP 111.222.32.80:52613;received=111.222.32.83;branch=z9hG4bK.0fa56909 From: "Klaus Darilion enum eyebea";tag=f1614469 To: ;tag=12345 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/anydomainxxx.at-081bd2d0", "SIP/klaus1|20") in new stack We're at 111.222.32.165 port 11756 Answering/Requesting with root capability 0x400 (ilbc) Answering with preferred capability 0x200 (speex) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 14 lines Reliably Transmitting (no NAT) to 111.222.33.3:5034: INVITE sip:klaus1@111.222.33.3:5034;line=6lfrccyr SIP/2.0 Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5c795c81;rport From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: Contact: Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Tue, 06 Sep 2005 08:46:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 320 v=0 o=root 30410 30410 IN IP4 111.222.32.165 s=session c=IN IP4 111.222.32.165 t=0 0 m=audio 11756 RTP/AVP 97 110 3 0 8 101 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called klaus1 -- B-channel 0/3 successfully restarted on span 1 samuel*CLI> <-- SIP read from 111.222.33.3:5034: REGISTER sip:asterisk-cvs.at SIP/2.0 Via: SIP/2.0/UDP 111.222.33.3:5034;branch=z9hG4bK-mo4b1ynijiwr;rport From: "klaus 371" ;tag=sxbbuesptq To: "klaus 371" Call-ID: 3c3f87c11a86-nty2lvqh17h7@83-136-33-3 CSeq: 3901 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56r P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 10.10.0.112 WWW-Contact: WWW-Contact: Expires: 60 Content-Length: 0 --- (17 headers 0 lines)--- Using latest request as basis request Sending to 111.222.33.3 : 5034 (non-NAT) Transmitting (no NAT) to 111.222.33.3:5034: SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.222.33.3:5034;branch=z9hG4bK-mo4b1ynijiwr From: "klaus 371" ;tag=sxbbuesptq To: "klaus 371" Call-ID: 3c3f87c11a86-nty2lvqh17h7@83-136-33-3 CSeq: 3901 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Transmitting (no NAT) to 111.222.33.3:5034: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 111.222.33.3:5034;branch=z9hG4bK-mo4b1ynijiwr From: "klaus 371" ;tag=sxbbuesptq To: "klaus 371" ;tag=as3d908c8b Call-ID: 3c3f87c11a86-nty2lvqh17h7@83-136-33-3 CSeq: 3901 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: WWW-Authenticate: Digest realm="asterisk", nonce="199c7feb" Content-Length: 0 --- Scheduling destruction of call '3c3f87c11a86-nty2lvqh17h7@83-136-33-3' in 15000 ms samuel*CLI> <-- SIP read from 111.222.33.3:5034: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5c795c81;rport=5060 From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/klaus1-54e9 is ringing Transmitting (no NAT) to 111.222.32.83:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.222.32.83;branch=z9hG4bK724e.58ff7932.0 Via: SIP/2.0/UDP 111.222.32.80:52613;received=111.222.32.83;branch=z9hG4bK.0fa56909 From: "Klaus Darilion enum eyebea";tag=f1614469 To: ;tag=12345 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- samuel*CLI> <-- SIP read from 111.222.33.3:5034: REGISTER sip:asterisk-cvs.at SIP/2.0 Via: SIP/2.0/UDP 111.222.33.3:5034;branch=z9hG4bK-eeohlc5gx44f;rport From: "klaus 371" ;tag=sxbbuesptq To: "klaus 371" Call-ID: 3c3f87c11a86-nty2lvqh17h7@83-136-33-3 CSeq: 3902 REGISTER Max-Forwards: 70 Contact: ;q=1.0 User-Agent: snom200-3.56r P-NAT-Refresh: 15 Supported: gruu Allow-Events: dialog X-Real-IP: 10.10.0.112 WWW-Contact: WWW-Contact: Authorization: Digest username="klaus1",realm="asterisk",nonce="199c7feb",uri="sip:asterisk-cvs.at",response="9de99d89f783a221784380fcdb8a8fdc",algorithm=md5 Expires: 60 Content-Length: 0 --- (18 headers 0 lines)--- Using latest request as basis request Sending to 111.222.33.3 : 5034 (non-NAT) Transmitting (no NAT) to 111.222.33.3:5034: SIP/2.0 100 Trying Via: SIP/2.0/UDP 111.222.33.3:5034;branch=z9hG4bK-eeohlc5gx44f From: "klaus 371" ;tag=sxbbuesptq To: "klaus 371" Call-ID: 3c3f87c11a86-nty2lvqh17h7@83-136-33-3 CSeq: 3902 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Saved useragent "snom200-3.56r" for peer klaus1 Transmitting (no NAT) to 111.222.33.3:5034: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.33.3:5034;branch=z9hG4bK-eeohlc5gx44f From: "klaus 371" ;tag=sxbbuesptq To: "klaus 371" ;tag=as3d908c8b Call-ID: 3c3f87c11a86-nty2lvqh17h7@83-136-33-3 CSeq: 3902 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Expires: 60 Contact: ;expires=60 Date: Tue, 06 Sep 2005 08:46:37 GMT Content-Length: 0 --- Scheduling destruction of call '3c3f87c11a86-nty2lvqh17h7@83-136-33-3' in 15000 ms samuel*CLI> <-- SIP read from 111.222.33.3:5034: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5c795c81;rport=5060 From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/klaus1-54e9 is ringing samuel*CLI> <-- SIP read from 111.222.33.3:5034: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5c795c81;rport=5060 From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/klaus1-54e9 is ringing samuel*CLI> <-- SIP read from 111.222.33.3:5034: SIP/2.0 200 Ok Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5c795c81;rport=5060 From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 102 INVITE Contact: User-Agent: snom200-3.56r Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Type: application/sdp Content-Length: 205 v=0 o=root 1270740617 1270740617 IN IP4 111.222.33.3 s=call c=IN IP4 111.222.33.3 t=0 0 m=audio 10094 RTP/AVP 3 101 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 10 lines)--- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 111.222.33.3:10094 Found description format gsm Found description format telephone-event Capabilities: us - 0x8060e (gsm|ulaw|alaw|speex|ilbc|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 111.222.33.3, port 5034 Transmitting (no NAT) to 111.222.33.3:5034: ACK sip:klaus1@111.222.33.3:5034;line=6lfrccyr SIP/2.0 Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK4e0ea258;rport From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Contact: Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/klaus1-54e9 answered SIP/anydomainxxx.at-081bd2d0 We're at 111.222.32.165 port 10028 Answering with preferred capability 0x400 (ilbc) Answering with preferred capability 0x200 (speex) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 111.222.32.83:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.32.83;branch=z9hG4bK724e.58ff7932.0 Via: SIP/2.0/UDP 111.222.32.80:52613;received=111.222.32.83;branch=z9hG4bK.0fa56909 Record-Route: From: "Klaus Darilion enum eyebea";tag=f1614469 To: ;tag=12345 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 320 v=0 o=root 30410 30410 IN IP4 111.222.32.165 s=session c=IN IP4 111.222.32.165 t=0 0 m=audio 10028 RTP/AVP 98 110 3 0 8 101 a=rtpmap:98 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/anydomainxxx.at-081bd2d0 and SIP/klaus1-54e9 samuel*CLI> <-- SIP read from 111.222.32.83:5060: ACK sip:371@111.222.32.165 SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 111.222.32.83;branch=0 Via: SIP/2.0/UDP 111.222.32.83;branch=0 Via: SIP/2.0/UDP 111.222.32.80:52613;received=111.222.32.83;branch=z9hG4bK.29e49379;rport=52614 Max-Forwards: 14 Contact: To: ;tag=12345 From: "Klaus Darilion enum eyebea";tag=f1614469 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 2 ACK Allow: ACK, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: sipsak Content-Length: 0 P-Behind-NAT: Yes --- (17 headers 0 lines)--- -- B-channel 0/4 successfully restarted on span 1 samuel*CLI> <-- SIP read from 111.222.32.83:5060: BYE sip:371@asterisk-cvs.at SIP/2.0 Record-Route: Via: SIP/2.0/UDP 111.222.32.83;branch=z9hG4bK824e.8e64bf7.0 Via: SIP/2.0/UDP 111.222.32.80:52618;received=111.222.32.83;branch=z9hG4bK.52bbe4c6;rport=52619 Max-Forwards: 15 Contact: To: ;tag=77777 From: "Klaus Darilion enum eyebea";tag=f1614469 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 3 BYE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: sipsak P-Behind-NAT: Yes --- (13 headers 0 lines)--- Sending to 111.222.32.83 : 5060 (non-NAT) Transmitting (no NAT) to 111.222.32.83:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.32.83;branch=z9hG4bK824e.8e64bf7.0 Via: SIP/2.0/UDP 111.222.32.80:52618;received=111.222.32.83;branch=z9hG4bK.52bbe4c6 Record-Route: From: "Klaus Darilion enum eyebea";tag=f1614469 To: ;tag=77777 Call-ID: 03614d34b41caf24@a2xhdXNwYw.. CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause:: Normal Clearing --- set_destination: Parsing for address/port to send to set_destination: set destination to 111.222.33.3, port 5034 Reliably Transmitting (no NAT) to 111.222.33.3:5034: BYE sip:klaus1@111.222.33.3:5034;line=6lfrccyr SIP/2.0 Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5e6cbf39;rport From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Contact: Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 103 BYE User-Agent: Asterisk PBX Content-Length: 0 --- samuel*CLI> <-- SIP read from 111.222.33.3:5034: SIP/2.0 200 OK Via: SIP/2.0/UDP 111.222.32.165:5060;branch=z9hG4bK5e6cbf39;rport=5060 From: "Klaus Darilion enum eyebea" ;tag=as3b1b4dcb To: ;tag=oeejhmfse9 Call-ID: 04498d2f76cc418434bd23e331cf3293@111.222.32.165 CSeq: 103 BYE Contact: User-Agent: snom200-3.56r Content-Length: 0 --- (9 headers 0 lines)--- Destroying call '04498d2f76cc418434bd23e331cf3293@111.222.32.165' Destroying call '03614d34b41caf24@a2xhdXNwYw..' -- B-channel 0/5 successfully restarted on span 1 -- B-channel 0/6 successfully restarted on span 1 Destroying call '3c3f87c11a86-nty2lvqh17h7@83-136-33-3' samuel*CLI> quit