Script started on Thu Aug 4 11:31:11 2005 obelix:~# 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 = 7797) 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: INVITE sip:86@voip.sysfrog.org;user=phone SIP/2.0 Record-Route: Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaa2.d426ee73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wkd3bnwrwnty;rport=6554 From: "u8" ;tag=nepqdd3qaq To: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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="42f1e0c40f4ef56f9709374107df72bd52c74f3e",uri="sip:86@voip.sysfrog.org;user=phone",response="4df61014ca2ae9626a5695df5b90a737",algorithm=md5 Content-Type: application/sdp Content-Length: 365 v=0 o=root 486573483 486573483 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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 - 3c27c49f09c4-lvi3sg7r98gp@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:7006 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=z9hG4bKaa2.d426ee73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wkd3bnwrwnty From: "u8" ;tag=nepqdd3qaq To: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Python("SIP/voip.sysfrog.org-081c7898", "incoming_sip_dial|86|voip.sysfrog.org|Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e0c40f4ef56f9709374107df72bd52c74f3e",uri="sip:86@voip.sysfrog.org;user=phone",response="4df61014ca2ae9626a5695df5b90a737",algorithm=md5|proxy") in new stack obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 17090 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u6@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK722d45f2 From: "u8" ;tag=as5173572d To: Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:31:55 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7845 7845 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 17090 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called u6@voip.sysfrog.org Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bKaa2.d426ee73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wkd3bnwrwnty From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK722d45f2 From: "u8" ;tag=as5173572d To: Call-ID: 4988431b611202d576a4331f2ed1f09b@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=117fd6a67486d1fbi0 From: "u8" ;tag=as5173572d Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK722d45f2 Record-Route: Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/voip.sysfrog.org-bf43 is ringing obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=117fd6a67486d1fbi0 From: "u8" ;tag=as5173572d Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK722d45f2 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=- 170587626 170587626 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7010 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:7010 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=z9hG4bK52c8e4fa Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-bf43 answered SIP/voip.sysfrog.org-081c7898 We're at ZZ.ZZ.ZZ.ZZ port 13742 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=z9hG4bKaa2.d426ee73.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wkd3bnwrwnty Record-Route: From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 7845 7845 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 13742 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-081c7898 and SIP/voip.sysfrog.org-bf43 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 17090 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=z9hG4bK4609a316 Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@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 7845 7846 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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 - - - - --- <-- 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=z9hG4bK4609a316 From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Call-ID: 4988431b611202d576a4331f2ed1f09b@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=3724 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=117fd6a67486d1fbi0 From: "u8" ;tag=as5173572d Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4609a316 Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170587832 170587832 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7010 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:7010 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=z9hG4bK3c683b7d Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@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-yv1wtvy80n5b;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 13742 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=z9hG4bK2b5cd620;rport Route: From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 7845 7846 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7010 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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=3728 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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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:7006 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) obelix*CLI> Aug 4 11:31:57 NOTICE[7814]: 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=z9hG4bK77f0e2f4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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:7006 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 11:31:58 NOTICE[7814]: 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=z9hG4bK76b7726f;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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:7006 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 11:31:59 NOTICE[7814]: 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=z9hG4bK0c237256;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bKba2.5e58dac4.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-v26nuj3ehr9o;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 3 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: 361 v=0 o=root 486573483 486573485 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10108 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 - 3c27c49f09c4-lvi3sg7r98gp@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 0.0.0.0:10108 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-bf43 We're at ZZ.ZZ.ZZ.ZZ port 13742 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=z9hG4bKba2.5e58dac4.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-v26nuj3ehr9o From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 7845 7847 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7010 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 11:32:00 NOTICE[7845]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-bf43 Opened file 0 '/var/lib/asterisk/moh-native/busstrafik' set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 17090 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u6@YY.YY.YY.YY:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d70724f Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@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 7845 7847 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 17090 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: 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-n0kad2gbxira;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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=z9hG4bK21981f45;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Retransmitting #1 (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=z9hG4bK5d70724f Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@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 7845 7847 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 17090 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=z9hG4bK5d70724f From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Call-ID: 4988431b611202d576a4331f2ed1f09b@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: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=117fd6a67486d1fbi0 From: "u8" ;tag=as5173572d Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5d70724f Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170588326 170588326 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7010 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:7010 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=z9hG4bK2b34065c Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 104 ACK User-Agent: Asterisk Content-Length: 0 --- 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=z9hG4bK0a7e.420ef872.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-5kho7kslxkmb;rport=6554 From: "u8" ;tag=s11kjz70yg To: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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="42f1e0cdea4b5839b69e29403dda5d29ccdb632c",uri="sip:85@voip.sysfrog.org;user=phone",response="98bb3e94f100d2c30982f85195312339",algorithm=md5 Content-Type: application/sdp Content-Length: 365 v=0 o=root 858138040 858138040 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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 - 3c27c4a809c4-tcazz5j3jkut@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:7014 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=z9hG4bK0a7e.420ef872.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-5kho7kslxkmb From: "u8" ;tag=s11kjz70yg To: Call-ID: 3c27c4a809c4-tcazz5j3jkut@snom190 CSeq: 2 INVITE User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- obelix*CLI> -- Executing Python("SIP/voip.sysfrog.org-081f9770", "incoming_sip_dial|85|voip.sysfrog.org|Digest username="u8",realm="voip.sysfrog.org",nonce="42f1e0cdea4b5839b69e29403dda5d29ccdb632c",uri="sip:85@voip.sysfrog.org;user=phone",response="98bb3e94f100d2c30982f85195312339",algorithm=md5|proxy") in new stack obelix*CLI> We're at ZZ.ZZ.ZZ.ZZ port 19766 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 12 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u5@voip.sysfrog.org SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK77a3ed71 From: "u8" ;tag=as0f114f64 To: Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE User-Agent: Asterisk Date: Thu, 04 Aug 2005 09:32:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 259 v=0 o=root 7849 7849 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 19766 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called u5@voip.sysfrog.org Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK0a7e.420ef872.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-5kho7kslxkmb From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK77a3ed71 From: "u8" ;tag=as0f114f64 To: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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: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=37f38a25cd4fb0fi0 From: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK77a3ed71 Record-Route: Server: Sipura/SPA841-3.1.2(d) Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/voip.sysfrog.org-1163 is ringing 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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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=z9hG4bK512a0799;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 To: ;tag=37f38a25cd4fb0fi0 From: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 102 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK77a3ed71 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=- 15783025 15783025 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7018 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:7018 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=z9hG4bK2558b43e Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- -- SIP/voip.sysfrog.org-1163 answered SIP/voip.sysfrog.org-081f9770 We're at ZZ.ZZ.ZZ.ZZ port 12870 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=z9hG4bK0a7e.420ef872.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-5kho7kslxkmb Record-Route: From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 7849 7849 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 12870 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-081f9770 and SIP/voip.sysfrog.org-1163 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 19766 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x1 (g723) Answering with capability 0x2 (gsm) Answering with capability 0x4 (ulaw) Answering with capability 0x10 (g726) Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 15 lines Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: INVITE sip:u5@YY.YY.YY.YY:6393 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3242d8f9 Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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 7849 7850 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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=z9hG4bK3242d8f9 From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Call-ID: 20fa873825b652202f7892fb4fcc77d4@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=37f38a25cd4fb0fi0 From: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 103 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3242d8f9 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15783115 15783115 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7018 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:7018 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=z9hG4bK084ac442 Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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-fuvzvkqgwbdh;rport=6554 From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 12870 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=z9hG4bK6c849115;rport Route: From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 7849 7850 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7018 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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=3723 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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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:7014 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=z9hG4bK74e56c77;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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:7014 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=z9hG4bK40797082;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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:7014 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=z9hG4bK50c6a7d4;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK1a7e.3e3c22c1.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8iieywber5m;rport=6554 From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138042 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10676 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 - 3c27c4a809c4-tcazz5j3jkut@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 0.0.0.0:10676 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-1163 We're at ZZ.ZZ.ZZ.ZZ port 12870 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=z9hG4bK1a7e.3e3c22c1.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-h8iieywber5m From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 7849 7851 IN IP4 YY.YY.YY.YY s=session c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7018 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 11:32:08 NOTICE[7849]: res_musiconhold.c:216 ast_moh_files_next: SIP/voip.sysfrog.org-1163 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 19766 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=z9hG4bK4e57ae34 Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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 7849 7851 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 19766 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=z9hG4bK4e57ae34 From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Call-ID: 20fa873825b652202f7892fb4fcc77d4@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=37f38a25cd4fb0fi0 From: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 104 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK4e57ae34 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15783345 15783345 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7018 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:7018 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=z9hG4bK64b73f3c Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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-yb3i658lk44f;rport=6554 From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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=z9hG4bK0324515c;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c27c4a809c4-tcazz5j3jkut%40snom190%3Bto-tag%3Das3624ada4%3Bfrom-tag%3Ds11kjz70yg 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-bf43 -- Stopped music on hold on SIP/voip.sysfrog.org-1163 Transmitting (no NAT) to YY.YY.YY.YY:6554: SIP/2.0 202 Accepted Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK39d800d4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK10ce5554;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Cobelix*CLI> ontact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 104 BYE User-Agent: Asterisk X-Asterisk-HangupCause:: Normal Clearing Content-Length: 0 --- set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 We're at ZZ.ZZ.ZZ.ZZ port 17090 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) 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=z9hG4bK34a52d7e Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@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 7845 7848 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 17090 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> set_destination: Parsing for address/port to send to 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=z9hG4bK34a52d7e From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Call-ID: 4988431b611202d576a4331f2ed1f09b@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=3723 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> 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=z9hG4bK25abaed8;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 19766 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=z9hG4bK3b8c3c18 Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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 7849 7852 IN IP4 ZZ.ZZ.ZZ.ZZ s=session c=IN IP4 ZZ.ZZ.ZZ.ZZ t=0 0 m=audio 19766 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> == Auto fallthrough, channel 'SIP/voip.sysfrog.org-081c7898' status is 'ANSWER' 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=z9hG4bK3b8c3c18 From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Call-ID: 20fa873825b652202f7892fb4fcc77d4@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=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=117fd6a67486d1fbi0 From: "u8" ;tag=as5173572d Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 105 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK34a52d7e Contact: u6 Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 238 Content-Type: application/sdp v=0 o=- 170589066 170589066 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7010 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:7010 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=z9hG4bK3e094931 Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 105 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=37f38a25cd4fb0fi0 From: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ CSeq: 105 INVITE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK3b8c3c18 Contact: u5 Server: Sipura/SPA841-3.1.2(d) Content-Length: 209 Content-Type: application/sdp v=0 o=- 15783541 15783541 IN IP4 YY.YY.YY.YY s=- c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7018 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:7018 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=z9hG4bK0e2d72f0 Route: From: "u8" ;tag=as0f114f64 To: ;tag=37f38a25cd4fb0fi0 Contact: Call-ID: 20fa873825b652202f7892fb4fcc77d4@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=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c27c4a809c4-tcazz5j3jkut%40snom190%3Bto-tag%3Das3624ada4%3Bfrom-tag%3Ds11kjz70yg 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 #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=z9hG4bK39d800d4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK10ce5554;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK25abaed8;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c27c4a809c4-tcazz5j3jkut%40snom190%3Bto-tag%3Das3624ada4%3Bfrom-tag%3Ds11kjz70yg 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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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=z9hG4bK3b9b467a;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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=z9hG4bK7e3a6543;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK39d800d4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK10ce5554;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK25abaed8;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@snom190 CSeq: 103 BYE 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=z9hG4bK39d800d4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK10ce5554;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK25abaed8;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c27c4a809c4-tcazz5j3jkut%40snom190%3Bto-tag%3Das3624ada4%3Bfrom-tag%3Ds11kjz70yg 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=z9hG4bK39d800d4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK10ce5554;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK25abaed8;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@snom190 CSeq: 103 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: BYE sip:asterisk@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK364a.af9d48b.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-fdddbb99 From: ;tag=37f38a25cd4fb0fi0 To: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@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) obelix*CLI> Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK364a.af9d48b.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6393;branch=z9hG4bK-fdddbb99 From: ;tag=37f38a25cd4fb0fi0 To: "u8" ;tag=as0f114f64 Call-ID: 20fa873825b652202f7892fb4fcc77d4@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 --- == Auto fallthrough, channel 'SIP/voip.sysfrog.org-bf43' status is 'ANSWER' obelix*CLI> set_destination: Parsing for address/port to send to set_destination: set destination to XX.XX.XX.XX, port 5060 Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: BYE sip:u6@YY.YY.YY.YY:6386 SIP/2.0 Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5ebedacf Route: From: "u8" ;tag=as5173572d To: ;tag=117fd6a67486d1fbi0 Contact: Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 106 BYE User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Destroying call '20fa873825b652202f7892fb4fcc77d4@ZZ.ZZ.ZZ.ZZ' obelix*CLI> <-- SIP read from XX.XX.XX.XX:5060: SIP/2.0 200 OK To: ;tag=117fd6a67486d1fbi0 From: "u8" ;tag=as5173572d Call-ID: 4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ CSeq: 106 BYE Via: SIP/2.0/UDP ZZ.ZZ.ZZ.ZZ:5060;branch=z9hG4bK5ebedacf Server: Sipura/SPA1001-2.0.13(SEg) Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '4988431b611202d576a4331f2ed1f09b@ZZ.ZZ.ZZ.ZZ' 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=z9hG4bK39d800d4;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK10ce5554;rport From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Contact: Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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=z9hG4bK25abaed8;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@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=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c27c4a809c4-tcazz5j3jkut%40snom190%3Bto-tag%3Das3624ada4%3Bfrom-tag%3Ds11kjz70yg 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=z9hG4bK6c849115;rport=5060 From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138041 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7014 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=z9hG4bK13d80786;rport From: ;tag=as3624ada4 To: "u8" ;tag=s11kjz70yg Contact: Call-ID: 3c27c4a809c4-tcazz5j3jkut@snom190 CSeq: 102 ACK User-Agent: Asterisk Content-Length: 0 --- obelix*CLI> Aug 4 11:32:21 WARNING[7814]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c27c49f09c4-lvi3sg7r98gp@snom190 for seqno 103 (Non-critical Request) Aug 4 11:32:21 WARNING[7814]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c27c49f09c4-lvi3sg7r98gp@snom190 for seqno 104 (Non-critical Request) Aug 4 11:32:21 WARNING[7814]: chan_sip.c:1055 retrans_pkt: Maximum retries exceeded on call 3c27c4a809c4-tcazz5j3jkut@snom190 for seqno 103 (Non-critical Request) Destroying call '3c27c4a809c4-tcazz5j3jkut@snom190' Destroying call '3c27c49f09c4-lvi3sg7r98gp@snom190' 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=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER Max-Forwards: 16 Contact: Refer-To: sip:85@ZZ.ZZ.ZZ.ZZ?Replaces=3c27c4a809c4-tcazz5j3jkut%40snom190%3Bto-tag%3Das3624ada4%3Bfrom-tag%3Ds11kjz70yg 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-default Transfer from u8 in incoming-sip-default Aug 4 11:32:24 NOTICE[7814]: chan_sip.c:5995 get_refer_info: Supervised transfer requested, but unable to find callid '3c27c4a809c4-tcazz5j3jkut@snom190'. Both legs must reside on Asterisk box to transfer at this time. Reliably Transmitting (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 --- obelix*CLI> Retransmitting #1 (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 obelix*CLI> --- obelix*CLI> Retransmitting #2 (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 obelix*CLI> --- 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=z9hG4bK2b5cd620;rport=5060 From: ;tag=as37eee92c To: "u8" ;tag=nepqdd3qaq Call-ID: 3c27c49f09c4-lvi3sg7r98gp@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 486573483 486573484 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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> Retransmitting #3 (no NAT) to XX.XX.XX.XX:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK8a2.7b289202.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-8uf4pd565nok From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 4 REFER User-Agent: Asterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Accept: application/sdp Content-Length: 0 obelix*CLI> --- 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=z9hG4bKe97e.8ddb09b3.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wpgamtgy94yy;rport=6554 From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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 858138040 858138043 IN IP4 10.122.32.84 s=call c=IN IP4 0.0.0.0 t=0 0 m=audio 10676 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 - 3c27c4a809c4-tcazz5j3jkut@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 0.0.0.0:10676 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=z9hG4bKe97e.8ddb09b3.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-wpgamtgy94yy From: "u8" ;tag=s11kjz70yg To: ;tag=as3624ada4 Call-ID: 3c27c4a809c4-tcazz5j3jkut@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: INVITE sip:86@ZZ.ZZ.ZZ.ZZ SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX;branch=z9hG4bK9a2.04fa7485.0 Via: SIP/2.0/UDP YY.YY.YY.YY:6554;branch=z9hG4bK-mtxoohyihtfw;rport=6554 From: "u8" ;tag=nepqdd3qaq To: ;tag=as37eee92c Call-ID: 3c27c49f09c4-lvi3sg7r98gp@snom190 CSeq: 5 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 486573483 486573486 IN IP4 YY.YY.YY.YY s=call c=IN IP4 YY.YY.YY.YY t=0 0 m=audio 7006 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 - 3c27c49f09c4-lvi3sg7r98gp@snom190 Sending to XX.XX.XX.XX : 5060 (non-NAT) Found peer 'sipsepp' obelix*CLI> -- Executing Python("SIP/voip.sysfrog.org-08201228", "incoming_sip_dial|85|ZZ.ZZ.ZZ.ZZ||proxy") in new stack obelix*CLI> quit obelix:~# Script done on Thu Aug 4 11:32:53 2005