Script started on Thu Aug 4 11:33:48 2005 obelix:~# gdb /usr/sbin/asterisk 7798obelix:~# asterisk -r Asterisk CVS-NHEAD-08/04/05-11:17:06, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= Connected to Asterisk CVS-NHEAD-08/04/05-11:17:06 currently running on obelix (pid = 7879) obelix*CLI> quitsip debug level 4set verbose 4 obelix*CLI> Verbosity was 0 and is now 4 obelix*CLI> set verbose 4quitsip debug level 4 obelix*CLI> Debugging level set to 4, file '' obelix*CLI> debug level 4set verbose 4quitsip debug obelix*CLI> 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=z9hG4bK0.b61c9346.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-gzdg7128fgp4;rport=6554 From: "u8" ;tag=6idlsapsga To: "u8" Call-ID: 3c26701cef42-r373exygo3mg@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: proxy-config Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e1598c321e92f90e8f98b565a6e4fb2c413d",uri="sip:u8@voip.sysfrog.org",response="7fb37115656148668f151e2e2d260e02",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=z9hG4bK0.b61c9346.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-gzdg7128fgp4 From: "u8" ;tag=6idlsapsga To: "u8" ;tag=as67fa5591 Call-ID: 3c26701cef42-r373exygo3mg@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 4 11:34:24 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c26701cef42-r373exygo3mg@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=z9hG4bK60ce.8d032ae3.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-2rrmk8hw28da;rport=6554 From: "u8" ;tag=uzvy8w9osj To: "u8" Call-ID: 3c26701d7530-dfh1jr3schcp@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: number-guessing Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e15aac0f3700863099e5323019b367fbace3",uri="sip:u8@voip.sysfrog.org",response="44bd48607cbaadb9391be051484e99cd",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=z9hG4bK60ce.8d032ae3.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-2rrmk8hw28da From: "u8" ;tag=uzvy8w9osj To: "u8" ;tag=as26f10ec2 Call-ID: 3c26701d7530-dfh1jr3schcp@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 4 11:34:25 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c26701d7530-dfh1jr3schcp@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=z9hG4bKcf31.07e24b.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz;rport=6554 From: "u8" ;tag=m7dxxmv5er To: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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="42f1e15aac0f3700863099e5323019b367fbace3",uri="sip:86@voip.sysfrog.org;user=phone",response="9b878c89a72b8d16d2909817cdb7019b",algorithm=md5 Content-Type: application/sdp Content-Length: 365 v=0 o=root 604740038 604740038 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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 - 3c26701d7ef4-avnd7zuk8gvc@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:7026 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=z9hG4bKcf31.07e24b.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz From: "u8" ;tag=m7dxxmv5er To: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Dial("SIP/voip.sysfrog.org-0823b5c0", "SIP/u6@voip.sysfrog.org|20|r") in new stack obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 12604 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> 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=z9hG4bK512440b0 From: "u8" ;tag=as2382a1f1 To: Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:34:25 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7915 7915 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 12604 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> -- Called u6@voip.sysfrog.org obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKcf31.07e24b.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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=z9hG4bK512440b0 From: "u8" ;tag=as2382a1f1 To: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@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=3724 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:6386 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=a075727231d1e08i0 From: "u8" ;tag=as2382a1f1 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK512440b0 Record-Route: Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (9 headers 0 lines)--- obelix*CLI> -- SIP/voip.sysfrog.org-643c is ringing obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=a075727231d1e08i0 From: "u8" ;tag=as2382a1f1 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK512440b0 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=- 170602628 170602628 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7030 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:7030 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71c19916 Route: From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> -- SIP/voip.sysfrog.org-643c answered SIP/voip.sysfrog.org-0823b5c0 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 15188 obelix*CLI> Answering with capability 0x8 (alaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 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=z9hG4bKcf31.07e24b.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-83phpcmzq2mz Record-Route: From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 7915 7915 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 15188 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> -- Attempting native bridge of SIP/voip.sysfrog.org-0823b5c0 and SIP/voip.sysfrog.org-643c obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 12604 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x1 (g723) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with capability 0x10 (g726) obelix*CLI> Answering with capability 0x100 (g729) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 15 lines obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u6@YY.YY.YY.YY:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d4f945d Route: From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 341 v=0 o=root 7915 7916 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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=z9hG4bK5d4f945d From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@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=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:6386 out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=a075727231d1e08i0 From: "u8" ;tag=as2382a1f1 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d4f945d Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170602831 170602831 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7030 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:7030 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4ca9d9a8 Route: From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@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:6554;branch=z9hG4bK-l4rgw428goso;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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 15188 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:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport Route: From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7915 7916 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7030 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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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:6554 out_uri=sip:u8@YY.YY.YY.YY:6554;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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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:7026 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 4 11:34:27 NOTICE[7896]: chan_sip.c:3353 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 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK771827bc;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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:7026 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 4 11:34:28 NOTICE[7896]: chan_sip.c:3353 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2a4119ff;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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:7026 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 4 11:34:29 NOTICE[7896]: chan_sip.c:3353 process_sdp: No compatible codecs! set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6e116267;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- 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=z9hG4bKdf31.d75a5361.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-p8axl569u7i7;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 361 v=0 o=root 604740038 604740040 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 - 3c26701d7ef4-avnd7zuk8gvc@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-643c We're at ZZ.ZZ.ZZ.ZZ port 15188 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=z9hG4bKdf31.d75a5361.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-p8axl569u7i7 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=root 7915 7917 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7030 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> Aug 4 11:34:31 NOTICE[7915]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-643c Opened file 0 '/var/lib/asterisk/moh-native/busstrafik' obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 12604 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u6@YY.YY.YY.YY:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71bb393a Route: From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7915 7917 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 12604 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=z9hG4bK71bb393a From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@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=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u6@YY.YY.YY.YY:6386 out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=a075727231d1e08i0 From: "u8" ;tag=as2382a1f1 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK71bb393a Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170603194 170603194 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7030 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:7030 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6dacd40a Route: From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 104 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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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)--- Response message INVITE arrived set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0550e83d;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 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:6554;branch=z9hG4bK-t2hw81oupfum;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: PUBLISH sip:u8@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9708.5817b935.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-9039x9hwm3x0;rport=6554 From: "u8" ;tag=p51it5v9zl To: "u8" Call-ID: 3c2670249c40-styma3r0wm8f@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: proxy-config Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e161f4c4775ff0cfbb65d0f7938f464b9651",uri="sip:u8@voip.sysfrog.org",response="3fac0588c5d6ed7b25e8ba91d29aeef8",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=z9hG4bK9708.5817b935.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-9039x9hwm3x0 From: "u8" ;tag=p51it5v9zl To: "u8" ;tag=as26f4bb0e Call-ID: 3c2670249c40-styma3r0wm8f@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 4 11:34:32 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c2670249c40-styma3r0wm8f@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=z9hG4bK850b.25b7e293.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-w5tblh4ai3eo;rport=6554 From: "u8" ;tag=pmcedhhmls To: "u8" Call-ID: 3c267025222e-k6gqu5vvg4t5@snom190 CSeq: 2 PUBLISH Max-Forwards: 16 Event: number-guessing Proxy-Authorization: Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e1625e45c56e4ebb391e81b776acb4140a3c",uri="sip:u8@voip.sysfrog.org",response="5b5fc85d7546e426bc123b6c7ee619fe",algorithm=md5 Content-Type: application/text Content-Length: 25 Number: 85 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=z9hG4bK850b.25b7e293.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-w5tblh4ai3eo From: "u8" ;tag=pmcedhhmls To: "u8" ;tag=as20ebbbe0 Call-ID: 3c267025222e-k6gqu5vvg4t5@snom190 CSeq: 2 PUBLISH User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- Aug 4 11:34:33 NOTICE[7896]: chan_sip.c:9714 handle_request: Unknown SIP command 'PUBLISH' from 'XX.XX.XX.XX' Destroying call '3c267025222e-k6gqu5vvg4t5@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=z9hG4bKe05d.accc3ec7.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v;rport=6554 From: "u8" ;tag=elvnjnhzkz To: Call-ID: 3c26702546cd-0z66kq5n4653@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="42f1e1625e45c56e4ebb391e81b776acb4140a3c",uri="sip:85@voip.sysfrog.org;user=phone",response="85e3a8cc5a6167b2eee77dd7998d2185",algorithm=md5 Content-Type: application/sdp Content-Length: 365 v=0 o=root 150881502 150881502 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 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 - 3c26702546cd-0z66kq5n4653@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:7034 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=z9hG4bKe05d.accc3ec7.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v From: "u8" ;tag=elvnjnhzkz To: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Dial("SIP/voip.sysfrog.org-08270198", "SIP/u5@voip.sysfrog.org|20|r") in new stack obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> 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=z9hG4bK2ac7dd81 From: "u8" ;tag=as7e51302d To: Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:34:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7918 7918 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18362 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> -- Called u5@voip.sysfrog.org obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKe05d.accc3ec7.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bK2ac7dd81 From: "u8" ;tag=as7e51302d To: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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=3728 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:6393 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=9e2ffb6524c174d3i0 From: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2ac7dd81 Record-Route: Server: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (9 headers 0 lines)--- obelix*CLI> -- SIP/voip.sysfrog.org-79a3 is ringing obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=9e2ffb6524c174d3i0 From: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2ac7dd81 Record-Route: Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 15797877 15797877 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7038 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7038 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK561da4fa Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> -- SIP/voip.sysfrog.org-79a3 answered SIP/voip.sysfrog.org-08270198 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 17206 obelix*CLI> Answering with capability 0x8 (alaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 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=z9hG4bKe05d.accc3ec7.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-ct010485ri6v Record-Route: From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 7918 7918 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 17206 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> -- Attempting native bridge of SIP/voip.sysfrog.org-08270198 and SIP/voip.sysfrog.org-79a3 obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x1 (g723) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with capability 0x10 (g726) obelix*CLI> Answering with capability 0x100 (g729) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 15 lines obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51c151c1 Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 341 v=0 o=root 7918 7919 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 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=z9hG4bK51c151c1 From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=9e2ffb6524c174d3i0 From: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK51c151c1 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15797942 15797942 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7038 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7038 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) 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:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7970b89c Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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: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:6554;branch=z9hG4bK-tijwvebsqu0l;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@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 17206 Answering with capability 0x8 (alaw) 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:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc0d441;rport Route: From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7918 7919 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7038 RTP/AVP 8 101 a=rtpmap:8 PCMA/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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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:6554 out_uri=sip:u8@YY.YY.YY.YY:6554;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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7034 Found description format pcma Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7fdeabd4;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7034 Found description format pcma Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3c446648;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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)--- Response message INVITE arrived set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK017953ca;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7034 Found description format pcma Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK26575cec;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bKf05d.4e2de167.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-0pp6ey27i376;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@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: 361 v=0 o=root 150881502 150881504 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 - 3c26702546cd-0z66kq5n4653@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 0.0.0.0:10704 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-79a3 We're at ZZ.ZZ.ZZ.ZZ port 17206 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=z9hG4bKf05d.4e2de167.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-0pp6ey27i376 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 3 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 213 v=0 o=root 7918 7920 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7038 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> Aug 4 11:34:35 NOTICE[7918]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-79a3 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik' obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7cd5bf6c Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7918 7920 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18362 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=z9hG4bK7cd5bf6c From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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=3722 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=9e2ffb6524c174d3i0 From: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7cd5bf6c Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15798137 15798137 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7038 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7038 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) 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:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK59c0c8bb Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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:6554;branch=z9hG4bK-bcye3dzwvidq;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Response message INVITE arrived set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0812b162;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 85 in incoming-sip Transfer from u8 in incoming-sip -- Stopped music on hold on SIP/voip.sysfrog.org-643c -- Stopped music on hold on SIP/voip.sysfrog.org-79a3 Transmitting (no NAT) to YY.YY.YY.YY:6554: SIP/2.0 202 Accepted Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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 6554 Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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 6554 Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060 obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18362 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5c6e92d4 Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 105 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7918 7921 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18362 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=z9hG4bK5c6e92d4 From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 105 INVITE 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:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=9e2ffb6524c174d3i0 From: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 105 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5c6e92d4 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15798344 15798344 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7038 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7038 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) 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:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK0cc463a1 Route: From: "u8" ;tag=as7e51302d To: ;tag=9e2ffb6524c174d3i0 Contact: Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ CSeq: 105 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 85 in incoming-sip Transfer from u8 in incoming-sip 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=z9hG4bK6877.862a0721.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-536332af From: ;tag=9e2ffb6524c174d3i0 To: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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=z9hG4bK6877.862a0721.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-536332af From: ;tag=9e2ffb6524c174d3i0 To: "u8" ;tag=as7e51302d Call-ID: 0d9aed4343b5ca9216867dcb1f813b38@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 --- obelix*CLI> set_destination: Parsing for address/port to send to obelix*CLI> set_destination: set destination to XX.XX.XX.XX, port 5060 obelix*CLI> Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: BYE sip:u6@YY.YY.YY.YY:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK448feee4 Route: From: "u8" ;tag=as2382a1f1 To: ;tag=a075727231d1e08i0 Contact: Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Destroying call '0d9aed4343b5ca9216867dcb1f813b38@ZZ.ZZ.ZZ.ZZ' obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=a075727231d1e08i0 From: "u8" ;tag=as2382a1f1 Call-ID: 18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ CSeq: 105 BYE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK448feee4 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '18f2bffd3fbdb9fd0bdfa76b0b8c00c1@ZZ.ZZ.ZZ.ZZ' obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 85 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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Response message INVITE arrived set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6c275e5a;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 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=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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)--- Response message INVITE arrived set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK78879740;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- Transfer to 85 in incoming-sip Transfer from u8 in incoming-sip obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK49230153;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK20a0d4fb;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- obelix*CLI> Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6fd8d4c6;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE 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=z9hG4bKc05d.770c3a73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 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: 361 v=0 o=root 150881502 150881505 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 - 3c26702546cd-0z66kq5n4653@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=z9hG4bKbf31.50e9de34.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 365 v=0 o=root 604740038 604740041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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 - 3c26701d7ef4-avnd7zuk8gvc@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=z9hG4bKc05d.770c3a73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 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: 361 v=0 o=root 150881502 150881505 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:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 365 v=0 o=root 604740038 604740041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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 <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (14 headers 11 lines)--- Response message INVITE arrived set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 obelix*CLI> Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7764249d;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Aug 4 11:34:49 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26701d7ef4-avnd7zuk8gvc@snom190 for seqno 103 (Non-critical Request) Aug 4 11:34:49 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26701d7ef4-avnd7zuk8gvc@snom190 for seqno 104 (Non-critical Request) Aug 4 11:34:49 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26702546cd-0z66kq5n4653@snom190 for seqno 103 (Non-critical Request) Destroying call '3c26702546cd-0z66kq5n4653@snom190' Destroying call '3c26701d7ef4-avnd7zuk8gvc@snom190' 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=z9hG4bKc05d.770c3a73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 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: 361 v=0 o=root 150881502 150881505 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 - 3c26702546cd-0z66kq5n4653@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 0.0.0.0:10704 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: Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Dial("SIP/voip.sysfrog.org-08242c08", "SIP/u5@voip.sysfrog.org|20|r") in new stack 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=z9hG4bKbf31.50e9de34.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 365 v=0 o=root 604740038 604740041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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 - 3c26701d7ef4-avnd7zuk8gvc@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:7026 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: Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 5 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Dial("SIP/voip.sysfrog.org-0824f240", "SIP/u6@voip.sysfrog.org|20|r") in new stack obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 11240 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> 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=z9hG4bK34d79d6c From: "u8" ;tag=as4276f4bc To: Contact: Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:34:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7925 7925 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 11240 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> -- Called u5@voip.sysfrog.org obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKc05d.770c3a73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 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=z9hG4bK34d79d6c From: "u8" ;tag=as4276f4bc To: Call-ID: 6abcb86a333a63890841e9a028d0241d@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:6393 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 18816 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> 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=z9hG4bK68cb5e8e From: "u8" ;tag=as6b06de5b To: Contact: Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:34:50 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7926 7926 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 18816 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> -- Called u6@voip.sysfrog.org obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKbf31.50e9de34.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 5 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=z9hG4bK68cb5e8e From: "u8" ;tag=as6b06de5b To: Call-ID: 2a995a615c74624f6dae97e16fef0a49@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=3728 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:6386 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=787170188283fba7i0 From: "u8" ;tag=as6b06de5b Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68cb5e8e Record-Route: Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (9 headers 0 lines)--- <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=a9d0f338a3ab2df1i0 From: "u8" ;tag=as4276f4bc Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34d79d6c Record-Route: Server: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (9 headers 0 lines)--- obelix*CLI> -- SIP/voip.sysfrog.org-2407 is ringing obelix*CLI> -- SIP/voip.sysfrog.org-a963 is ringing obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=a9d0f338a3ab2df1i0 From: "u8" ;tag=as4276f4bc Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34d79d6c Record-Route: Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Content-Type: application/sdp v=0 o=- 15799626 15799626 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7040 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Transmitting (no NAT) to XX.XX.XX.XX:5060: ACK sip:u5@YY.YY.YY.YY:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK09f63201 Route: From: "u8" ;tag=as4276f4bc To: ;tag=a9d0f338a3ab2df1i0 Contact: Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-2407 answered SIP/voip.sysfrog.org-08242c08 We're at ZZ.ZZ.ZZ.ZZ port 19156 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=z9hG4bKc05d.770c3a73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8pwkg5bzsfdk From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 7925 7925 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 19156 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-08242c08 and SIP/voip.sysfrog.org-2407 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 11240 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:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK15d2ebd4 Route: From: "u8" ;tag=as4276f4bc To: ;tag=a9d0f338a3ab2df1i0 Contact: Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7925 7926 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 11240 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 aobelix*CLI> =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=z9hG4bK15d2ebd4 From: "u8" ;tag=as4276f4bc To: ;tag=a9d0f338a3ab2df1i0 Call-ID: 6abcb86a333a63890841e9a028d0241d@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=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u5@YY.YY.YY.YY:6393 out_uri=sip:u5@YY.YY.YY.YY:6393 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:6554;branch=z9hG4bK-wi073saxgi79;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 4 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 YY.YY.YY.YY, port 6554 We're at ZZ.ZZ.ZZ.ZZ port 19156 Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 10 lines Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7925 7926 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/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/2.0 200 OK To: ;tag=a9d0f338a3ab2df1i0 From: "u8" ;tag=as4276f4bc Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK15d2ebd4 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15800261 15800261 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7040 Found description format PCMA Found description format telephone-event Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) 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:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68a03ff2 Route: From: "u8" ;tag=as4276f4bc To: ;tag=a9d0f338a3ab2df1i0 Contact: Call-ID: 6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ CSeq: 103 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7925 7926 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/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/2.0 200 Ok Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2200acbc;rport=5060 From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 217 v=0 o=root 604740038 604740039 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=787170188283fba7i0 From: "u8" ;tag=as6b06de5b Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK68cb5e8e 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=- 170605152 170605152 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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:7042 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7b3b896d Route: From: "u8" ;tag=as6b06de5b To: ;tag=787170188283fba7i0 Contact: Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-a963 answered SIP/voip.sysfrog.org-0824f240 We're at ZZ.ZZ.ZZ.ZZ port 14658 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=z9hG4bKbf31.50e9de34.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mif2jgu3epfq From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 5 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 212 v=0 o=root 7926 7926 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 14658 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-0824f240 and SIP/voip.sysfrog.org-a963 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 18816 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba Route: From: "u8" ;tag=as6b06de5b To: ;tag=787170188283fba7i0 Contact: Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 341 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7026 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=z9hG4bK1a8ae2ba From: "u8" ;tag=as6b06de5b To: ;tag=787170188283fba7i0 Call-ID: 2a995a615c74624f6dae97e16fef0a49@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:u6@YY.YY.YY.YY:6386 out_uri=sip:u6@YY.YY.YY.YY:6386 via_cnt==1" --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=787170188283fba7i0 From: "u8" ;tag=as6b06de5b Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170606152 170606152 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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:7042 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4d5dd139 Route: From: "u8" ;tag=as6b06de5b To: ;tag=787170188283fba7i0 Contact: Call-ID: 2a995a615c74624f6dae97e16fef0a49@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:6554;branch=z9hG4bK-10fz744gtbti;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 5 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 YY.YY.YY.YY, port 6554 We're at ZZ.ZZ.ZZ.ZZ port 14658 Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 11 headers, 10 lines Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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/2.0 200 OK To: ;tag=787170188283fba7i0 From: "u8" ;tag=as6b06de5b Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a8ae2ba Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170606152 170606152 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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:7042 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:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2e69a5b3 Route: From: "u8" ;tag=as6b06de5b To: ;tag=787170188283fba7i0 Contact: Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 103 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7925 7926 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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: BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8ef6.edf59345.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6386;branch=z9hG4bK-1bc36695 From: ;tag=787170188283fba7i0 To: "u8" ;tag=as6b06de5b Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 101 BYE Max-Forwards: 16 User-Agent: Sipura/SPA1001-2.0.13(SEg) 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=z9hG4bK8ef6.edf59345.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6386;branch=z9hG4bK-1bc36695 From: ;tag=787170188283fba7i0 To: "u8" ;tag=as6b06de5b Call-ID: 2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ CSeq: 101 BYE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 Xobelix*CLI> -Asterisk-HangupCause:: Normal Clearing --- obelix*CLI> Destroying call '2a995a615c74624f6dae97e16fef0a49@ZZ.ZZ.ZZ.ZZ' obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7925 7926 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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: REFER sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaf31.400cdb23.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8pd5nb80aof;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c26702546cd-0z66kq5n4653%40snom190%3Bto-tag%3Das67271bcd%3Bfrom-tag%3Delvnjnhzkz Referred-By: sip:u8@voip.sysfrog.org User-Agent: snom190/3.60k Content-Length: 0 --- (13 headers 0 lines)--- obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK1a229502;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7925 7926 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7040 RTP/AVP 8 101 a=rtpmap:8 PCMA/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: BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK401f.c19573f6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-5dc97018 From: ;tag=a9d0f338a3ab2df1i0 To: "u8" ;tag=as4276f4bc Call-ID: 6abcb86a333a63890841e9a028d0241d@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=z9hG4bK401f.c19573f6.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-5dc97018 From: ;tag=a9d0f338a3ab2df1i0 To: "u8" ;tag=as4276f4bc Call-ID: 6abcb86a333a63890841e9a028d0241d@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 --- obelix*CLI> Destroying call '6abcb86a333a63890841e9a028d0241d@ZZ.ZZ.ZZ.ZZ' 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=z9hG4bK0cc0d441;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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: 217 v=0 o=root 150881502 150881503 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 RTP/AVP 8 101 a=rtpmap:8 pcma/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 8 Found RTP audio format 101 Peer audio RTP is at port YY.YY.YY.YY:7034 Found description format pcma Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Transmitting (no NAT) to YY.YY.YY.YY:6554: ACK sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7aa3156c;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@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 6554 Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bKd05d.1e2d0254.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 5 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er 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 Aug 4 11:35:08 WARNING[7896]: chan_sip.c:8933 attempt_transfer: Transfer attempted without dual ownership? Transmitting (no NAT) to YY.YY.YY.YY:6554: SIP/2.0 202 Accepted Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 5 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to YY.YY.YY.YY, port 6554 Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 104 NOTIFY User-Agent: Asterisk Event: refer;id=5 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 6554 Reliably Transmitting (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #4 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 5 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er 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> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #1 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 104 NOTIFY User-Agent: Asterisk Event: refer;id=5 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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #5 (no NAT) to YY.YY.YY.YY:6554: INVITE sip:u8@YY.YY.YY.YY:6554 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK66aa23ef;rport From: ;tag=as6d3df7db To: "u8" ;tag=m7dxxmv5er Contact: Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 102 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 213 v=0 o=root 7926 7927 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7042 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: REFER sip:85@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 5 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er 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> Retransmitting #3 (no NAT) to YY.YY.YY.YY:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #2 (no NAT) to YY.YY.YY.YY:6554: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 104 NOTIFY User-Agent: Asterisk Event: refer;id=5 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:6554: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 105 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Aug 4 11:35:12 WARNING[7896]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c26701d7ef4-avnd7zuk8gvc@snom190 for seqno 102 (Non-critical Request) Destroying call '3c26701d7ef4-avnd7zuk8gvc@snom190' 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=z9hG4bK8f31.d4457802.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-jm4qpgkrrysj;rport=6554 From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@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: 361 v=0 o=root 604740038 604740042 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 - 3c26701d7ef4-avnd7zuk8gvc@snom190 obelix*CLI> Sending to XX.XX.XX.XX : 5060 (non-NAT) obelix*CLI> Found peer 'sipsepp' obelix*CLI> Found RTP audio format 0 obelix*CLI> Found RTP audio format 8 obelix*CLI> Found RTP audio format 9 obelix*CLI> Found RTP audio format 2 obelix*CLI> Found RTP audio format 3 obelix*CLI> Found RTP audio format 18 obelix*CLI> Found RTP audio format 4 obelix*CLI> Found RTP audio format 101 obelix*CLI> Peer audio RTP is at port 0.0.0.0:10996 obelix*CLI> Found description format pcmu obelix*CLI> Found description format pcma obelix*CLI> Found description format g722 obelix*CLI> Found description format g726-32 obelix*CLI> Found description format gsm obelix*CLI> Found description format g729 obelix*CLI> Found description format g723 obelix*CLI> Found description format telephone-event obelix*CLI> Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw) obelix*CLI> Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) obelix*CLI> Looking for 86 in incoming-sip obelix*CLI> list_route: hop: obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8f31.d4457802.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-jm4qpgkrrysj From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 6 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Dial("SIP/voip.sysfrog.org-0824d6b0", "SIP/u6@voip.sysfrog.org|20|r") in new stack 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=z9hG4bKa05d.e51aacd.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-nx2g5ux53lzs;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 6 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp obelix*CLI> 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: 365 v=0 o=root 150881502 150881506 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 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 obelix*CLI> --- (18 headers 17 lines)--- obelix*CLI> Using INVITE request as basis request - 3c26702546cd-0z66kq5n4653@snom190 obelix*CLI> Sending to XX.XX.XX.XX : 5060 (non-NAT) obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 11728 obelix*CLI> Answering/Requesting with root capability 0x8 (alaw) obelix*CLI> Answering with capability 0x2 (gsm) obelix*CLI> Answering with capability 0x4 (ulaw) obelix*CLI> Answering with non-codec capability 0x1 (telephone-event) obelix*CLI> 12 headers, 12 lines obelix*CLI> 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=z9hG4bK6c3d9c1b From: "u8" ;tag=as510099ca To: Contact: Call-ID: 2035685651fcc9842b9d68995d5717c3@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:35:13 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7935 7935 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 11728 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> -- Called u6@voip.sysfrog.org obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8f31.d4457802.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-jm4qpgkrrysj From: "u8" ;tag=m7dxxmv5er To: ;tag=as6d3df7db Call-ID: 3c26701d7ef4-avnd7zuk8gvc@snom190 CSeq: 6 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=z9hG4bK6c3d9c1b From: "u8" ;tag=as510099ca To: Call-ID: 2035685651fcc9842b9d68995d5717c3@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=3724 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:6386 via_cnt==1" obelix*CLI> --- (9 headers 0 lines)--- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 180 Ringing To: ;tag=592c53bf95be9d09i0 From: "u8" ;tag=as510099ca Call-ID: 2035685651fcc9842b9d68995d5717c3@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK6c3d9c1b Record-Route: Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (9 headers 0 lines)--- obelix*CLI> -- SIP/voip.sysfrog.org-a359 is ringing obelix*CLI> Retransmitting #4 (no NAT) to XX.XX.XX.XX:5060: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK2d8e2315;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 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=z9hG4bK2d8e2315;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 103 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:6554;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- Response message BYE arrived 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=z9hG4bKa05d.e51aacd.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-nx2g5ux53lzs;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 6 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp obelix*CLI> 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: 365 v=0 o=root 150881502 150881506 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 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> Retransmitting #3 (no NAT) to XX.XX.XX.XX:5060: NOTIFY sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK63b571f0;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 104 NOTIFY User-Agent: Asterisk Event: refer;id=5 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 XX.XX.XX.XX:5060: BYE sip:u8@YY.YY.YY.YY:6554;line=c371zoiv SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3343430d;rport From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Contact: Call-ID: 3c26702546cd-0z66kq5n4653@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=z9hG4bK3343430d;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@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=3728 req_src_ip=ZZ.ZZ.ZZ.ZZ req_src_port=5060 in_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- Response message BYE arrived <-- 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=z9hG4bK63b571f0;rport=5060 From: ;tag=as67271bcd To: "u8" ;tag=elvnjnhzkz Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 104 NOTIFY 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:6554;line=c371zoiv out_uri=sip:u8@YY.YY.YY.YY:6554;line=c371zoiv via_cnt==1" --- (9 headers 0 lines)--- Response message NOTIFY arrived Destroying call '3c26702546cd-0z66kq5n4653@snom190' 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=z9hG4bKd05d.1e2d0254.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 5 REFER Max-Forwards: 16 Contact: Refer-To: sip:86@ZZ.ZZ.ZZ.ZZ?Replaces=3c26701d7ef4-avnd7zuk8gvc%40snom190%3Bto-tag%3Das6d3df7db%3Bfrom-tag%3Dm7dxxmv5er 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-default Transfer from u8 in incoming-sip-default Aug 4 11:35:14 WARNING[7896]: chan_sip.c:8933 attempt_transfer: Transfer attempted without dual ownership? Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 202 Accepted Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKd05d.1e2d0254.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-7jcuire0rolz From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 5 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Aug 4 11:35:14 NOTICE[7896]: chan_sip.c:3518 copy_header: No field 'Call-ID' present to copy 11 headers, 1 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: NOTIFY SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK485b4484;rport From: ;tag=as02127b3c To: Contact: CSeq: 102 NOTIFY User-Agent: Asterisk Event: refer;id=5 Subscription-state: terminated;reason=noresource Content-Type: message/sipfrag;version=2.0 Content-Length: 14 SIP/2.0 200 OK --- Aug 4 11:35:14 NOTICE[7896]: chan_sip.c:3518 copy_header: No field 'Call-ID' present to copy Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: BYE SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK7afa5be8;rport From: ;tag=as02127b3c To: ;tag=as02127b3c Contact: CSeq: 103 BYE 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=z9hG4bKa05d.e51aacd.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-nx2g5ux53lzs;rport=6554 From: "u8" ;tag=elvnjnhzkz To: ;tag=as67271bcd Call-ID: 3c26702546cd-0z66kq5n4653@snom190 CSeq: 6 INVITE Max-Forwards: 16 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60k Accept: application/sdp obelix*CLI> 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: 365 v=0 o=root 150881502 150881506 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7034 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 - 3c26702546cd-0z66kq5n4653@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) Found peer 'sipsepp' obelix*CLI> == Primary D-Channel on span 1 up == Primary D-Channel on span 1 up obelix*CLI> == Primary D-Channel on span 1 up obelix*CLI> quit obelix:~# Script done on Thu Aug 4 11:35:35 2005