To: Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 1 INVITE Max-Forwards: 70 Contact: P-Key-Flags: keys="3" User-Agent: snom190/3.60i 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: 364 v=0 o=root 1184052172 1184052172 IN IP4 10.60.33.9 s=call c=IN IP4 10.60.33.9 t=0 0 m=audio 60380 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 --- (17 headers 17 lines)--- Using INVITE request as basis request - 3c41ee87a393-sdu3blgcu8bo@snom190 Sending to 10.60.33.9 : 2051 (NAT) Found user '309' 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 10.60.33.9:60380 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 - 0xc (ulaw|alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 0400518501 in from-internal (domain 10.60.33.102) list_route: hop: Transmitting (no NAT) to 10.60.33.9:2051: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-v5320lemhqjc;rport;received=10.60.33.9 From: "Darren" ;tag=bgybpgubi1 To: Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Executing Macro("SIP/309-5000", "dialout-trunk|4|0400518501||") in new stack -- Executing GotoIf("SIP/309-5000", "1?3:2") in new stack -- Goto (macro-dialout-trunk,s,3) -- Executing Macro("SIP/309-5000", "user-callerid") in new stack -- Executing GotoIf("SIP/309-5000", "0?report") in new stack -- Executing GotoIf("SIP/309-5000", "0?start") in new stack -- Executing Set("SIP/309-5000", "REALCALLERIDNUM=309") in new stack -- Executing NoOp("SIP/309-5000", "REALCALLERIDNUM is 309") in new stack -- Executing Set("SIP/309-5000", "AMPUSER=309") in new stack -- Executing Set("SIP/309-5000", "AMPUSERCIDNAME=Darren") in new stack -- Executing GotoIf("SIP/309-5000", "0?report") in new stack -- Executing Set("SIP/309-5000", "CALLERID(all)=Darren <309>") in new stack -- Executing NoOp("SIP/309-5000", "Using CallerID "Darren" <309>") in new stack -- Executing Macro("SIP/309-5000", "record-enable|309|OUT") in new stack -- Executing GotoIf("SIP/309-5000", "0 > 0?2:4") in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI("SIP/309-5000", "recordingcheck|20060702-130714|1151816834.76") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060702-130714|1151816834.76: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp("SIP/309-5000", "No recording needed") in new stack -- Executing Macro("SIP/309-5000", "outbound-callerid|4") in new stack -- Executing GotoIf("SIP/309-5000", "1?start") in new stack -- Goto (macro-outbound-callerid,s,3) -- Executing NoOp("SIP/309-5000", "REALCALLERIDNUM is 309") in new stack -- Executing Set("SIP/309-5000", "USEROUTCID=") in new stack -- Executing Set("SIP/309-5000", "EMERGENCYCID=") in new stack -- Executing Set("SIP/309-5000", "TRUNKOUTCID=") in new stack -- Executing GotoIf("SIP/309-5000", "1?trunkcid") in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing GotoIf("SIP/309-5000", "1?usercid") in new stack -- Goto (macro-outbound-callerid,s,13) -- Executing GotoIf("SIP/309-5000", "1?report") in new stack -- Goto (macro-outbound-callerid,s,15) -- Executing NoOp("SIP/309-5000", "CallerID set to "Darren" <309>") in new stack -- Executing Set("SIP/309-5000", "GROUP()=OUT_4") in new stack -- Executing GotoIf("SIP/309-5000", "0?108") in new stack -- Executing Set("SIP/309-5000", "DIAL_NUMBER=0400518501") in new stack -- Executing Set("SIP/309-5000", "DIAL_TRUNK=4") in new stack -- Executing AGI("SIP/309-5000", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/309-5000", "OUTNUM=0400518501") in new stack -- Executing Set("SIP/309-5000", "custom=ZAP/4") in new stack -- Executing GotoIf("SIP/309-5000", "0?16") in new stack -- Executing Dial("SIP/309-5000", "ZAP/4/0400518501|120|r") in new stack -- Called 4/0400518501 Transmitting (no NAT) to 10.60.33.9:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-v5320lemhqjc;rport;received=10.60.33.9 From: "Darren" ;tag=bgybpgubi1 To: ;tag=as59bed056 Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 --- -- Zap/4-1 answered SIP/309-5000 We're at 10.60.33.102 port 16944 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 10.60.33.9:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-v5320lemhqjc;rport;received=10.60.33.9 From: "Darren" ;tag=bgybpgubi1 To: ;tag=as59bed056 Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Type: application/sdp Content-Length: 240 v=0 o=root 26441 26441 IN IP4 10.60.33.102 s=session c=IN IP4 10.60.33.102 t=0 0 m=audio 16944 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- trixboxdhn*CLI> <-- SIP read from 10.60.33.9:2051: ACK sip:0400518501@10.60.33.102 SIP/2.0 Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-uexurcnm06eu;rport From: "Darren" ;tag=bgybpgubi1 To: ;tag=as59bed056 Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 1 ACK Max-Forwards: 70 Contact: Content-Length: 0 --- (9 headers 0 lines)--- trixboxdhn*CLI> <-- SIP read from 10.60.33.9:2051: BYE sip:0400518501@10.60.33.102 SIP/2.0 Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-dsicr5hmakou;rport From: "Darren" ;tag=bgybpgubi1 To: ;tag=as59bed056 Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 2 BYE Max-Forwards: 70 Contact: User-Agent: snom190/3.60i Content-Length: 0 --- (10 headers 0 lines)--- Sending to 10.60.33.9 : 2051 (NAT) Transmitting (NAT) to 10.60.33.9:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 10.60.33.9:2051;branch=z9hG4bK-dsicr5hmakou;rport;received=10.60.33.9 From: "Darren" ;tag=bgybpgubi1 To: ;tag=as59bed056 Call-ID: 3c41ee87a393-sdu3blgcu8bo@snom190 CSeq: 2 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause: Normal Clearing trixboxdhn*CLI> --- -- Hungup 'Zap/4-1' Destroying call '3c41ee87a393-sdu3blgcu8bo@snom190' trixboxdhn*CLI>