obelix:~# asterisk -r Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-HEAD currently running on obelix (pid = 21669) obelix*CLI> set verbose 4 Verbosity was 0 and is now 4 obelix*CLI> debug level 4 Debugging level set to 4, file '' obelix*CLI> sip debug SIP Debugging enabled obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: PUBLISH sip:u8@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK90ca.2a8a95e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-crninnpv72tp;rport=2051 From: "u8" ;tag=zp1e1ubab5 To: "u8" Call-ID: 3c26705a35b6-da4v6b16wgno@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: proxy-config Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:u8@voip.sysfrog.org",response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",algorithm=md5 Content-Type: application/text Content-Length: 0 --- (12 headers 0 lines)--- Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK90ca.2a8a95e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-crninnpv72tp From: "u8" ;tag=zp1e1ubab5 To: "u8" ;tag=as1e054265 Call-ID: 3c26705a35b6-da4v6b16wgno@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 3 11:20:01 NOTICE[21685]: chan_sip.c:9703 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c26705a35b6-da4v6b16wgno@snom190' obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:85@voip.sysfrog.org;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK3772.cccb4411.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-8geeyt3kg63v;rport=2051 From: "u8" ;tag=rmqzsyd1i5 To: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 2 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:85@voip.sysfrog.org;user=phone",response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",algorithm=md5 Content-Type: application/sdp Content-Length: 367 v=0 o=root 1483266644 1483266644 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (20 headers 17 lines)--- Using INVITE request as basis request - 3c26705aa122-fz0pe55yfpm3@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) Found peer 'sipsepp' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2948 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/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 XX.XX.XX.XX:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK3772.cccb4411.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-8geeyt3kg63v From: "u8" ;tag=rmqzsyd1i5 To: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/voip.sysfrog.org-08235d90", "SIP/u5@voip.sysfrog.org|20|r") in new stack We're at ZZ.ZZ.ZZ.ZZ port 18498 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 XX.XX.XX.XX:5060: INVITE sip:u5@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK11ab3eec From: "u8" ;tag=as41d061ef To: Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Wed, 03 Aug 2005 09:20:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 21707 21707 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18498 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 Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK3772.cccb4411.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-8geeyt3kg63v From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK11ab3eec From: "u8" ;tag=as41d061ef To: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@voip.sysfrog.org out_uri=sip:u5@YY.YY.YY.YY:1032 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=54dae1502b0b2619i0 From: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK11ab3eec Record-Route: Server: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/voip.sysfrog.org-0449 is ringing obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=54dae1502b0b2619i0 From: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK11ab3eec Record-Route: Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 207 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 7070890 7070890 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2952 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 YY.YY.YY.YY:2952 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 XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u5@YY.YY.YY.YY:1032 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK417d83d3 Route: From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-0449 answered SIP/voip.sysfrog.org-08235d90 We're at ZZ.ZZ.ZZ.ZZ port 18372 Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK3772.cccb4411.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-8geeyt3kg63v Record-Route: From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 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 21707 21707 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18372 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-08235d90 and SIP/voip.sysfrog.org-0449 set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 18498 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x1 (g723) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x10 (g726) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 15 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u5@YY.YY.YY.YY:1032 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2b63fd65 Route: From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 343 v=0 o=root 21707 21708 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 8 4 3 0 111 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2b63fd65 From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:1032 out_uri=sip:u5@YY.YY.YY.YY:1032 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: ACK sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-v5b0bcolk5u0;rport=2051 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 2 ACK Max-Forwards: 16 Contact: Content-Length: 0 --- (10 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 18372 Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u8@YY.YY.YY.YY:1056 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport Route: From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 215 v=0 o=root 21707 21708 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2952 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3724 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:1056 out_uri=sip:u8@YY.YY.YY.YY:1056;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=54dae1502b0b2619i0 From: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2b63fd65 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 207 Content-Type: application/sdp v=0 o=- 7070962 7070962 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2952 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 --- (10 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2952 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) set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u5@YY.YY.YY.YY:1032 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6d148449 Route: From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 103 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=54dae1502b0b2619i0 From: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2b63fd65 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 207 Content-Type: application/sdp v=0 o=- 7070962 7070962 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2952 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 --- (10 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2952 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) set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u5@YY.YY.YY.YY:1032 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4622dd26 Route: From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 103 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1483266644 1483266645 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2948 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:03 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2afef731;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1483266644 1483266645 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2948 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:03 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6e69c83b;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1483266644 1483266645 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2948 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:04 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK237d72a3;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK4772.655d3bd4.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-ow8nxxwxvuvo;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 3 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 363 v=0 o=root 1483266644 1483266646 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10996 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendonly --- (18 headers 17 lines)--- Using INVITE request as basis request - 3c26705aa122-fz0pe55yfpm3@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 0.0.0.0:10996 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/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-0449 We're at ZZ.ZZ.ZZ.ZZ port 18372 Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK4772.655d3bd4.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-ow8nxxwxvuvo From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 215 v=0 o=root 21707 21709 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2952 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- Aug 3 11:20:05 NOTICE[21707]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-0449 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik' set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 18498 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 XX.XX.XX.XX:5060: INVITE sip:u5@YY.YY.YY.YY:1032 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK765e016a Route: From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 21707 21709 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18498 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 - - - - --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK765e016a From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3723 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:1032 out_uri=sip:u5@YY.YY.YY.YY:1032 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=54dae1502b0b2619i0 From: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK765e016a Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 207 Content-Type: application/sdp v=0 o=- 7071242 7071242 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2952 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 --- (10 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2952 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) set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u5@YY.YY.YY.YY:1032 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK14c28005 Route: From: "u8" ;tag=as41d061ef To: ;tag=54dae1502b0b2619i0 Contact: Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 104 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: ACK sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-hgpn3kly9wzl;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 3 ACK Max-Forwards: 16 Contact: Content-Length: 0 --- (10 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1483266644 1483266645 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- !!!!!!!---------------************* Why are we here with this packet???? INVITE Response message is INVITE set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5773a714;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: PUBLISH sip:u8@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9581.9578ecb.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-icfqb0s6ub0c;rport=1056 From: "u8" ;tag=tva6y9e3cx To: "u8" Call-ID: 3c26705f46cd-gk0psh6of12u@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: proxy-config Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:u8@voip.sysfrog.org",response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",algorithm=md5 Content-Type: application/text Content-Length: 0 --- (12 headers 0 lines)--- Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9581.9578ecb.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-icfqb0s6ub0c From: "u8" ;tag=tva6y9e3cx To: "u8" ;tag=as4b7f2ee1 Call-ID: 3c26705f46cd-gk0psh6of12u@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 3 11:20:06 NOTICE[21685]: chan_sip.c:9703 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c26705f46cd-gk0psh6of12u@snom190' obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: PUBLISH sip:u8@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK7195.5e4762c2.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rntsj5j6iwof;rport=1056 From: "u8" ;tag=otulfd6y5e To: "u8" Call-ID: 3c26705fc0df-2ifdybphcs73@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: number-guessing Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:u8@voip.sysfrog.org",response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",algorithm=md5 Content-Type: application/text Content-Length: 25 Number: 86 Max-Hits: 3 --- (12 headers 2 lines)--- Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK7195.5e4762c2.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rntsj5j6iwof From: "u8" ;tag=otulfd6y5e To: "u8" ;tag=as3a308e7e Call-ID: 3c26705fc0df-2ifdybphcs73@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 3 11:20:07 NOTICE[21685]: chan_sip.c:9703 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c26705fc0df-2ifdybphcs73@snom190' obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:86@voip.sysfrog.org;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf497.11f39197.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-pbzgqhvxjzfj;rport=1056 From: "u8" ;tag=sejeff6bqv To: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 2 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",uri="sip:86@voip.sysfrog.org;user=phone",response="XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX",algorithm=md5 Content-Type: application/sdp Content-Length: 367 v=0 o=root 1030674848 1030674848 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (20 headers 17 lines)--- Using INVITE request as basis request - 3c26705fe30d-k15qab6fif6i@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) Found peer 'sipsepp' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2956 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 86 in incoming-sip list_route: hop: list_route: hop: Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf497.11f39197.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-pbzgqhvxjzfj From: "u8" ;tag=sejeff6bqv To: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/voip.sysfrog.org-08263e88", "SIP/u6@voip.sysfrog.org|20|r") in new stack We're at ZZ.ZZ.ZZ.ZZ port 13010 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 XX.XX.XX.XX:5060: INVITE sip:u6@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK07e70001 From: "u8" ;tag=as0c50d46b To: Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Wed, 03 Aug 2005 09:20:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 21710 21710 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 13010 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 u6@voip.sysfrog.org Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf497.11f39197.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-pbzgqhvxjzfj From: "u8" ;tag=sejeff6bqv To: ;tag=as5d6d8f40 Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK07e70001 From: "u8" ;tag=as0c50d46b To: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@voip.sysfrog.org out_uri=sip:u6@YY.YY.YY.YY:1031 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=41042d512f8090fi0 From: "u8" ;tag=as0c50d46b Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK07e70001 Record-Route: Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/voip.sysfrog.org-8bcd is ringing obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=41042d512f8090fi0 From: "u8" ;tag=as0c50d46b Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK07e70001 Record-Route: Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura Content-Type: application/sdp v=0 o=- 161876909 161876909 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2960 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (13 headers 12 lines)--- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2960 Found description format PCMU Found description format NSE 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 XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u6@YY.YY.YY.YY:1031 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK671bd0cb Route: From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-8bcd answered SIP/voip.sysfrog.org-08263e88 We're at ZZ.ZZ.ZZ.ZZ port 15212 Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf497.11f39197.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-pbzgqhvxjzfj Record-Route: From: "u8" ;tag=sejeff6bqv To: ;tag=as5d6d8f40 Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 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 21710 21710 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 15212 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-08263e88 and SIP/voip.sysfrog.org-8bcd set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 13010 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x1 (g723) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x10 (g726) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 15 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u6@YY.YY.YY.YY:1031 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK39b31a1c Route: From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 343 v=0 o=root 21710 21711 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 8 4 3 0 111 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK39b31a1c From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3723 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:1031 out_uri=sip:u6@YY.YY.YY.YY:1031 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=41042d512f8090fi0 From: "u8" ;tag=as0c50d46b Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK39b31a1c Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 161877048 161877048 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2960 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 12 lines)--- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2960 Found description format PCMU Found description format NSE 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) set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u6@YY.YY.YY.YY:1031 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK64f4fdd5 Route: From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 103 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: ACK sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-aa8x9kda1wyk;rport=1056 From: "u8" ;tag=sejeff6bqv To: ;tag=as5d6d8f40 Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 2 ACK Max-Forwards: 16 Contact: Content-Length: 0 --- (10 headers 0 lines)--- set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 15212 Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u8@YY.YY.YY.YY:1056 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport Route: From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 215 v=0 o=root 21710 21711 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2960 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:1056 out_uri=sip:u8@YY.YY.YY.YY:1056;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1030674848 1030674849 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2956 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:09 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4f20858f;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1030674848 1030674849 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2956 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:09 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1e76380f;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1030674848 1030674849 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2956 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:10 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK278b7b70;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1483266644 1483266645 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- !!!!!!!---------------************* Why are we here with this packet???? INVITE Response message is INVITE set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5bb7dc3c;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK1772.356b46e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rw92q2jzmgu7;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26705fe30d-k15qab6fif6i%40snom190%3Bto-tag%3Das5d6d8f40%3Bfrom-tag%3Dsejeff6bqv Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 86 in incoming-sip Transfer from u8 in incoming-sip -- Stopped music on hold on SIP/voip.sysfrog.org-0449 Transmitting (no NAT) to YY.YY.YY.YY:1056: SIP/2.0 202 Accepted Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK1772.356b46e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rw92q2jzmgu7 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 4 REFER User-Agent: Asterisk 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 YY.YY.YY.YY, port 1056 Reliably Transmitting (no NAT) to YY.YY.YY.YY:1056: NOTIFY sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5e6778ca;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=4 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 YY.YY.YY.YY, port 1056 Reliably Transmitting (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4aa937;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Reliably Transmitting (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0c6b9b76;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 13010 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 XX.XX.XX.XX:5060: INVITE sip:u6@YY.YY.YY.YY:1031 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5a939c18 Route: From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 261 v=0 o=root 21710 21712 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 13010 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 - - - - --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5a939c18 From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3723 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:1031 out_uri=sip:u6@YY.YY.YY.YY:1031 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=41042d512f8090fi0 From: "u8" ;tag=as0c50d46b Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5a939c18 Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 161877358 161877358 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2960 RTP/AVP 0 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:100 NSE/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv --- (10 headers 12 lines)--- Found RTP audio format 0 Found RTP audio format 100 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2960 Found description format PCMU Found description format NSE 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) set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u6@YY.YY.YY.YY:1031 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2abd5e73 Route: From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 104 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK1772.356b46e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rw92q2jzmgu7;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26705fe30d-k15qab6fif6i%40snom190%3Bto-tag%3Das5d6d8f40%3Bfrom-tag%3Dsejeff6bqv Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 86 in incoming-sip Transfer from u8 in incoming-sip obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1030674848 1030674849 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2956 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:12 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK579f7889;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Reliably Transmitting (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #1 (no NAT) to YY.YY.YY.YY:1056: NOTIFY sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5e6778ca;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Retransmitting #1 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4aa937;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- Retransmitting #1 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0c6b9b76;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK1772.356b46e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rw92q2jzmgu7;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26705fe30d-k15qab6fif6i%40snom190%3Bto-tag%3Das5d6d8f40%3Bfrom-tag%3Dsejeff6bqv Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 86 in incoming-sip Transfer from u8 in incoming-sip Retransmitting #1 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf9c4.522a8043.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1032;branch=z9hG4bK-94a5b1a6 From: ;tag=54dae1502b0b2619i0 To: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 101 BYE Max-Forwards: 16 User-Agent: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (10 headers 0 lines)--- Sending to XX.XX.XX.XX : 5060 (non-NAT) Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf9c4.522a8043.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1032;branch=z9hG4bK-94a5b1a6 From: ;tag=54dae1502b0b2619i0 To: "u8" ;tag=as41d061ef Call-ID: 7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ CSeq: 101 BYE User-Agent: Asterisk 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 XX.XX.XX.XX, port 5060 Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: BYE sip:u6@YY.YY.YY.YY:1031 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK67210fd8 Route: From: "u8" ;tag=as0c50d46b To: ;tag=41042d512f8090fi0 Contact: Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=41042d512f8090fi0 From: "u8" ;tag=as0c50d46b Call-ID: 4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ CSeq: 105 BYE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK67210fd8 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '4e83e099539a12a064fcd3c251c242a0@ZZ.ZZ.ZZ.ZZ' Destroying call '7cbfc5d810e1f19b127eafbb79b2bb1e@ZZ.ZZ.ZZ.ZZ' Retransmitting #2 (no NAT) to YY.YY.YY.YY:1056: NOTIFY sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5e6778ca;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Retransmitting #2 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4aa937;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- Retransmitting #2 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0c6b9b76;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #2 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0e65c9c8;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1030674848 1030674849 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2956 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:2956 Found description format pcmu Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Aug 3 11:20:16 NOTICE[21685]: chan_sip.c:3344 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK508b2a1c;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Reliably Transmitting (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51a53239;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #3 (no NAT) to YY.YY.YY.YY:1056: NOTIFY sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5e6778ca;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Retransmitting #3 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4aa937;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- Retransmitting #3 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0c6b9b76;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK1772.356b46e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rw92q2jzmgu7;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26705fe30d-k15qab6fif6i%40snom190%3Bto-tag%3Das5d6d8f40%3Bfrom-tag%3Dsejeff6bqv Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 86 in incoming-sip Transfer from u8 in incoming-sip Retransmitting #3 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #1 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51a53239;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK25c47918;rport=5060 From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 INVITE Contact: Session-Expires: 3600 User-Agent: snom190/3.60k 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: 219 v=0 o=root 1483266644 1483266645 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 101 a=rtpmap:0 pcmu/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- !!!!!!!---------------************* Why are we here with this packet???? INVITE Response message is INVITE set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 1056 Transmitting (no NAT) to YY.YY.YY.YY:1056: ACK sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK041719e5;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- Retransmitting #4 (no NAT) to YY.YY.YY.YY:1056: NOTIFY sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5e6778ca;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Retransmitting #4 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4aa937;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- Retransmitting #4 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0c6b9b76;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #4 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #2 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51a53239;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- Retransmitting #5 (no NAT) to YY.YY.YY.YY:1056: NOTIFY sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5e6778ca;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 103 NOTIFY User-Agent: Asterisk Event: refer;id=4 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Retransmitting #5 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4aa937;rport From: ;tag=as539d35fd To: "u8" ;tag=rmqzsyd1i5 Contact: Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- Retransmitting #5 (no NAT) to YY.YY.YY.YY:1056: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0c6b9b76;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK1772.356b46e6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rw92q2jzmgu7;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26705fe30d-k15qab6fif6i%40snom190%3Bto-tag%3Das5d6d8f40%3Bfrom-tag%3Dsejeff6bqv Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 86 in incoming-sip Transfer from u8 in incoming-sip obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK0597.c3112b51.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-lnebkxeahnt5;rport=1056 From: "u8" ;tag=sejeff6bqv To: ;tag=as5d6d8f40 Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 3 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 363 v=0 o=root 1030674848 1030674850 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10704 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendonly --- (18 headers 17 lines)--- Using INVITE request as basis request - 3c26705fe30d-k15qab6fif6i@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK2772.a975234.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rm97vcz1pdho;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 5 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 367 v=0 o=root 1483266644 1483266647 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (18 headers 17 lines)--- Using INVITE request as basis request - 3c26705aa122-fz0pe55yfpm3@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) Retransmitting #5 (no NAT) to XX.XX.XX.XX:5060: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 480 User Not Registered Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK23423b33;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 104 BYE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:1056;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:1056;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- !!!!!!!---------------************* Why are we here with this packet???? BYE Response message is BYE Retransmitting #3 (no NAT) to XX.XX.XX.XX:5060: BYE sip:u8@YY.YY.YY.YY:1056;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51a53239;rport From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Contact: Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 480 User Not Registered Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51a53239;rport=5060 From: ;tag=as5d6d8f40 To: "u8" ;tag=sejeff6bqv Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 105 BYE Server: OpenSer (0.9.4 (i386/linux)) Content-Length: 0 Warning: 392 XX.XX.XX.XX:5060 "Noisy feedback tells: pid=3723 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:1056;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:1056;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- !!!!!!!---------------************* Why are we here with this packet???? BYE Response message is BYE obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK0597.c3112b51.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-lnebkxeahnt5;rport=1056 From: "u8" ;tag=sejeff6bqv To: ;tag=as5d6d8f40 Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 3 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 363 v=0 o=root 1030674848 1030674850 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10704 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendonly --- (18 headers 17 lines)--- Ignoring this INVITE request <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK2772.a975234.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-rm97vcz1pdho;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 5 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 367 v=0 o=root 1483266644 1483266647 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 2948 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (18 headers 17 lines)--- Ignoring this INVITE request obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKf672.caa028e.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-bg9palzv8eqc;rport=1056 From: "u8" ;tag=rmqzsyd1i5 To: ;tag=as539d35fd Call-ID: 3c26705aa122-fz0pe55yfpm3@snom190 CSeq: 6 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 363 v=0 o=root 1483266644 1483266648 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10996 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendonly --- (18 headers 17 lines)--- Using INVITE request as basis request - 3c26705aa122-fz0pe55yfpm3@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd497.a71246a5.0 Via: SIP/2.0/UDP YY.YY.YY.YY:1056;branch=z9hG4bK-bpncc01nmae2;rport=1056 From: "u8" ;tag=sejeff6bqv To: ;tag=as5d6d8f40 Call-ID: 3c26705fe30d-k15qab6fif6i@snom190 CSeq: 4 INVITE Max-Forwards: 16 Contact: