Connected to Asterisk 1.6.0-beta8 currently running on mainast (pid = 22046) Verbosity is at least 3 mainast*CLI> sip set debug ip 192.168.0.10 SIP Debugging Enabled for IP: 192.168.0.10 <--- SIP read from UDP://192.168.0.10:5062 ---> INVITE sip:2XXXXX0@192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5062;branch=z9hG4bK3d1f2972;rport Max-Forwards: 70 From: "42XXXXX0" ;tag=as2357da72 To: Contact: Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 INVITE User-Agent: MFCR2 PSTN Gateway Date: Fri, 30 May 2008 04:47:56 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Content-Type: application/sdp Content-Length: 284 v=0 o=root 852424615 852424615 IN IP4 192.168.0.10 s=Asterisk PBX SVN--r c=IN IP4 192.168.0.10 t=0 0 m=audio 26968 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <-------------> --- (14 headers 13 lines) --- == Using SIP RTP CoS mark 5 == Using SIP VRTP CoS mark 6 Sending to 192.168.0.10 : 5062 (NAT) Using INVITE request as basis request - 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 No user '042XXXXX0' in SIP users list Found peer 'trunk-asterisk' for '042XXXXX0' from 192.168.0.10:5062 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 101 Peer audio RTP is at port 192.168.0.10:26968 Found audio description format PCMA for ID 8 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - 0x5c000e (gsm|ulaw|alaw|h261|h263|h263p|mpeg4), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 192.168.0.10:26968 Peer video RTP is at port 192.168.0.10:13625 Looking for 2XXXXX0 in from_pstn (domain 192.168.0.10) list_route: hop: mainast*CLI> <--- Transmitting (no NAT) to 192.168.0.10:5062 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.10:5062;branch=z9hG4bK3d1f2972;received=192.168.0.10;rport=5062 From: "42XXXXX0" ;tag=as2357da72 To: Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 INVITE User-Agent: Main Asterisk Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> -- Executing [2XXXXX0@from_pstn:1] Answer("SIP/192.168.0.10-b6b00da0", "") in new stack Audio is at 192.168.0.10 port 12404 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Reliably Transmitting (no NAT) to 192.168.0.10:5062 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5062;branch=z9hG4bK3d1f2972;received=192.168.0.10;rport=5062 From: "42XXXXX0" ;tag=as2357da72 To: ;tag=as32cfed06 Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 INVITE User-Agent: Main Asterisk Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 289 v=0 o=root 515652342 515652342 IN IP4 192.168.0.10 s=Asterisk PBX 1.6.0-beta8 c=IN IP4 192.168.0.10 t=0 0 m=audio 12404 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv <------------> -- Executing [2XXXXX0@from_pstn:2] Set("SIP/192.168.0.10-b6b00da0", "CALLERID(number)=9042XXXXX0") in new stack -- Executing [2XXXXX0@from_pstn:3] Wait("SIP/192.168.0.10-b6b00da0", "0.75") in new stack mainast*CLI> <--- SIP read from UDP://192.168.0.10:5062 ---> ACK sip:2XXXXX0@192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5062;branch=z9hG4bK37eb1136;rport Max-Forwards: 70 From: "42XXXXX0" ;tag=as2357da72 To: ;tag=as32cfed06 Contact: Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 ACK User-Agent: MFCR2 PSTN Gateway Content-Length: 0 <-------------> --- (10 headers 0 lines) --- mainast*CLI> -- Executing [2XXXXX0@from_pstn:4] Goto("SIP/192.168.0.10-b6b00da0", "hostess,s,1") in new stack -- Goto (hostess,s,1) -- Executing [s@hostess:1] BackGround("SIP/192.168.0.10-b6b00da0", "gracias-por-llamar") in new stack -- Playing 'gracias-por-llamar.slin' (language 'es') mainast*CLI> mainast*CLI> mainast*CLI> mainast*CLI> == CDR updated on SIP/192.168.0.10-b6b00da0 -- Executing [4083@hostess:1] Answer("SIP/192.168.0.10-b6b00da0", "") in new stack -- Executing [4083@hostess:2] Playback("SIP/192.168.0.10-b6b00da0", "tt-weasels") in new stack -- Playing 'tt-weasels.gsm' (language 'es') mainast*CLI> mainast*CLI> mainast*CLI> -- Executing [4083@hostess:3] Hangup("SIP/192.168.0.10-b6b00da0", "") in new stack == Spawn extension (hostess, 4083, 3) exited non-zero on 'SIP/192.168.0.10-b6b00da0' Scheduling destruction of SIP dialog '29e343e93e884a0049f09c68776d3c0e@192.168.0.10' in 32000 ms (Method: ACK) set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.0.10, port 5060 Reliably Transmitting (no NAT) to 192.168.0.10:5060: BYE sip:042XXXXX0@192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK271f9c2b;rport Max-Forwards: 70 From: ;tag=as32cfed06 To: "42XXXXX0" ;tag=as2357da72 Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 BYE User-Agent: Main Asterisk Server Content-Length: 0 --- mainast*CLI> <--- SIP read from UDP://192.168.0.10:5060 ---> BYE sip:042XXXXX0@192.168.0.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK271f9c2b;rport Max-Forwards: 70 From: ;tag=as32cfed06 To: "42XXXXX0" ;tag=as2357da72 Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 BYE User-Agent: Main Asterisk Server Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Sending to 192.168.0.10 : 5060 (NAT) Scheduling destruction of SIP dialog '29e343e93e884a0049f09c68776d3c0e@192.168.0.10' in 32000 ms (Method: BYE) <--- Transmitting (NAT) to 192.168.0.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK271f9c2b;received=192.168.0.10;rport=5060 From: ;tag=as32cfed06 To: "42XXXXX0" ;tag=as2357da72 Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 BYE User-Agent: Main Asterisk Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <------------> <--- SIP read from UDP://192.168.0.10:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.10:5060;branch=z9hG4bK271f9c2b;received=192.168.0.10;rport=5060 From: ;tag=as32cfed06 To: "42XXXXX0" ;tag=as2357da72 Call-ID: 29e343e93e884a0049f09c68776d3c0e@192.168.0.10 CSeq: 102 BYE User-Agent: Main Asterisk Server Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces, timer Contact: Content-Length: 0 <-------------> --- (11 headers 0 lines) --- SIP Response message for INCOMING dialog BYE arrived Really destroying SIP dialog '29e343e93e884a0049f09c68776d3c0e@192.168.0.10' Method: BYE