*CLI> sip set debug ip 192.168.0.141 SIP Debugging Enabled for IP: 192.168.0.141 *CLI> -- Accepting UNAUTHENTICATED call from 192.168.0.77: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (), > priority = mine -- Executing [s@from-iax:1] NoOp("IAX2/192.168.0.77:4569-2", "appel ext") in new stack -- Executing [s@from-iax:2] Dial("IAX2/192.168.0.77:4569-2", "Local/6666@local-extensions") in new stack -- Executing [6666@local-extensions:1] NoOp("Local/6666@local-extensions-9fbb,2", "################ENTREE QUEUE") in new stack -- Executing [6666@local-extensions:2] Queue("Local/6666@local-extensions-9fbb,2", "Standard|rTt|||15") in new stack -- Called 6666@local-extensions -- Local/6666@local-extensions-9fbb,1 is ringing -- outgoing agentcall, to agent '5100', on 'Local/100@local-extensions-8ba9,1' -- Executing [100@local-extensions:1] Dial("Local/100@local-extensions-8ba9,2", "SIP/100|30|Ttr") in new stack Audio is at 192.168.0.147 port 11948 Adding codec 0x2 (gsm) to SDP Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.0.141:2051: INVITE sip:100@192.168.0.141:2051;line=k6fmz7p1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK47ad50d0;rport Max-Forwards: 70 From: "102" ;tag=as3028fe8a To: Contact: Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 INVITE User-Agent: Asterisk PBX SVN-trunk-r56209 Date: Thu, 01 Mar 2007 17:17:30 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 289 v=0 o=root 28336 28336 IN IP4 192.168.0.147 s=session c=IN IP4 192.168.0.147 t=0 0 m=audio 11948 RTP/AVP 3 0 8 101 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- -- Called 100 -- Agent/5100 is ringing <--- SIP read from 192.168.0.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK47ad50d0;rport=5060 From: "102" ;tag=as3028fe8a To: ;tag=ulaf2dak86 Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/100-081c9598 is ringing -- Agent/5100 is ringing <--- SIP read from 192.168.0.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK47ad50d0;rport=5060 From: "102" ;tag=as3028fe8a To: ;tag=ulaf2dak86 Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/100-081c9598 is ringing <--- SIP read from 192.168.0.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK47ad50d0;rport=5060 From: "102" ;tag=as3028fe8a To: ;tag=ulaf2dak86 Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/100-081c9598 is ringing <--- SIP read from 192.168.0.141:2051 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK47ad50d0;rport=5060 From: "102" ;tag=as3028fe8a To: ;tag=ulaf2dak86 Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 INVITE Contact: ;flow-id=1 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- SIP/100-081c9598 is ringing <--- SIP read from 192.168.0.141:2051 ---> SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK47ad50d0;rport=5060 From: "102" ;tag=as3028fe8a To: ;tag=ulaf2dak86 Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 INVITE Contact: ;flow-id=1 User-Agent: snom320/6.5.2 Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces, callerid Content-Type: application/sdp Content-Length: 219 v=0 o=root 931995695 931995696 IN IP4 192.168.0.141 s=call c=IN IP4 192.168.0.141 t=0 0 m=audio 52018 RTP/AVP 3 101 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> --- (13 headers 11 lines) --- Found RTP audio format 3 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.141:52018 Found description format gsm for ID 3 Found description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x2 (gsm)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x2 (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.141:52018 --- set_address_from_contact host '192.168.0.141' list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.141, port 2051 Transmitting (no NAT) to 192.168.0.141:2051: ACK sip:100@192.168.0.141:2051;line=k6fmz7p1 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.147:5060;branch=z9hG4bK722b4528;rport Max-Forwards: 70 From: "102" ;tag=as3028fe8a To: ;tag=ulaf2dak86 Contact: Call-ID: 7a52181656c6c8d13ab27a5333fa198e@192.168.0.147 CSeq: 102 ACK User-Agent: Asterisk PBX SVN-trunk-r56209 Content-Length: 0 --- -- Call on SIP/100-081c9598 left from hold -- SIP/100-081c9598 answered Local/100@local-extensions-8ba9,2 -- Agent/5100 answered Local/6666@local-extensions-9fbb,2 -- Local/6666@local-extensions-9fbb,1 answered IAX2/192.168.0.77:4569-2 == Spawn extension (local-extensions, 100, 1) exited non-zero on 'Local/100@local-extensions-8ba9,2' -- Executing [h@local-extensions:1] Hangup("Local/100@local-extensions-8ba9,2", "") in new stack == Spawn extension (local-extensions, h, 1) exited non-zero on 'Local/100@local-extensions-8ba9,2' == Spawn extension (local-extensions, 6666, 2) exited non-zero on 'Local/6666@local-extensions-9fbb,2' -- Executing [h@local-extensions:1] Hangup("Local/6666@local-extensions-9fbb,2", "") in new stack == Spawn extension (local-extensions, h, 1) exited non-zero on 'Local/6666@local-extensions-9fbb,2' *CLI> *CLI> core show channels concise SIP/100-081c9598!local-extensions!s!1!Up!(None)!!100!!3!-!Local/6666@local-extensions-9fbb,1 Agent/5100!local-extensions!!1!Up!Bridged Call!IAX2/192.168.0.77:4569-2!102!!3!-!IAX2/192.168.0.77:4569-2 IAX2/192.168.0.77:4569-2!from-iax!s!2!Up!Dial!Local/6666@local-extensions!102!!3!12!SIP/100-081c9598 *CLI> core show channels concise SIP/100-081c9598!local-extensions!s!1!Up!(None)!!100!!3!-!Local/6666@local-extensions-9fbb,1 Agent/5100!local-extensions!!1!Up!Bridged Call!IAX2/192.168.0.77:4569-2!102!!3!-!IAX2/192.168.0.77:4569-2 IAX2/192.168.0.77:4569-2!from-iax!s!2!Up!Dial!Local/6666@local-extensions!102!!3!15!SIP/100-081c9598 *CLI> Segmentation fault (core dumped)