-- /var/log/asterisk/debug -- [Jan 15 16:11:56] DEBUG[11852] channel.c: Driver for channel 'SIP/user-b1e25258' does not support indication 3, emulating it [Jan 15 16:11:58] DEBUG[11852] pbx.c: Function result is '3103161650' [Jan 15 16:11:58] DEBUG[11852] pbx.c: Function result is '3103161650' [Jan 15 16:11:58] DEBUG[11852] pbx.c: Function result is '3103161650' [Jan 15 16:11:58] DEBUG[11852] pbx.c: Function result is 'REDONDO, CA' [Jan 15 16:11:58] DEBUG[11852] pbx.c: Expression result is '1' [Jan 15 16:11:59] DEBUG[11852] pbx.c: Oooh, got something to jump out with ('1')! -- console output -- -- Executing [3109533338@incoming:1] Answer("SIP/user-b1e25258", "1") in new stack -- Executing [3109533338@incoming:2] Ringing("SIP/user-b1e25258", "") in new stack -- Executing [3109533338@incoming:3] Wait("SIP/user-b1e25258", "2") in new stack -- Executing [3109533338@incoming:4] NoOp("SIP/user-b1e25258", "Incoming name 3103161650") in new stack -- Executing [3109533338@incoming:5] NoOp("SIP/user-b1e25258", "Incoming number 3103161650") in new stack -- Executing [3109533338@incoming:6] AGI("SIP/user-b1e25258", "callerid_name_lookup.agi|3103161650") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/callerid_name_lookup.agi -- AGI Script callerid_name_lookup.agi completed, returning 0 -- Executing [3109533338@incoming:7] NoOp("SIP/user-b1e25258", "AGI Returned REDONDO, CA") in new stack -- Executing [3109533338@incoming:8] Set("SIP/user-b1e25258", "CALLERID(name)=REDONDO, CA") in new stack -- Executing [3109533338@incoming:9] NoOp("SIP/user-b1e25258", "Name is set to REDONDO, CA") in new stack -- Executing [3109533338@incoming:10] GotoIf("SIP/user-b1e25258", "1?greeting|s|1") in new stack -- Goto (greeting,s,1) -- Executing [s@greeting:1] BackGround("SIP/user-b1e25258", "omnis-gsm/thankyouforcalling") in new stack -- Playing 'omnis-gsm/thankyouforcalling' (language 'en') == CDR updated on SIP/user-b1e25258 -- Executing [1@greeting:1] Macro("SIP/user-b1e25258", "queueit|customerservice") in new stack -- Executing [s@macro-queueit:1] Set("SIP/user-b1e25258", "__MONITOR_FILENAME=/calls/1168906316.1") in new stack -- Executing [s@macro-queueit:2] Set("SIP/user-b1e25258", "__MONITOR_EXEC_ARGS=1168906316.1") in new stack -- Executing [s@macro-queueit:3] Queue("SIP/user-b1e25258", "customerservice|i") in new stack -- Started music on hold, class 'default', on channel 'SIP/user-b1e25258' Audio is at 216.239.128.31 port 15906 Adding codec 0x4 (ulaw) to SDP Adding codec 0x2 (gsm) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 216.239.128.199:5060: INVITE sip:3201@216.239.128.199 SIP/2.0 Via: SIP/2.0/UDP 216.239.128.31:5060;branch=z9hG4bK7364e328;rport From: "REDONDO, CA" ;tag=as4f62c931 To: Contact: Call-ID: 3082416d51ef8b8a3c7283a00fb4cfff@216.239.128.31 CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 16 Jan 2007 00:11:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 291 v=0 o=root 11449 11449 IN IP4 216.239.128.31 s=session c=IN IP4 216.239.128.31 t=0 0 m=audio 15906 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/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 --- vidar*CLI> <--- SIP read from 216.239.128.199:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 216.239.128.31:5060;branch=z9hG4bK7364e328;rport From: "REDONDO, CA" ;tag=as4f62c931 To: ;tag=75EFD617-C86ACDA2 CSeq: 102 INVITE Call-ID: 3082416d51ef8b8a3c7283a00fb4cfff@216.239.128.31 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131 Content-Length: 0 <-------------> --- (9 headers 0 lines) --- vidar*CLI> <--- SIP read from 216.239.128.199:5060 ---> SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 216.239.128.31:5060;branch=z9hG4bK7364e328;rport From: "REDONDO, CA" ;tag=as4f62c931 To: ;tag=75EFD617-C86ACDA2 CSeq: 102 INVITE Call-ID: 3082416d51ef8b8a3c7283a00fb4cfff@216.239.128.31 Contact: User-Agent: PolycomSoundPointIP-SPIP_650-UA/2.0.3.0131 Diversion: ;reason="deflection" Content-Length: 0 <-------------> --- (10 headers 0 lines) --- -- Got SIP response 302 "Moved Temporarily" back from 216.239.128.199 Transmitting (no NAT) to 216.239.128.199:5060: ACK sip:3201@216.239.128.199 SIP/2.0 Via: SIP/2.0/UDP 216.239.128.31:5060;branch=z9hG4bK7364e328;rport From: "REDONDO, CA" ;tag=as4f62c931 To: ;tag=75EFD617-C86ACDA2 Contact: Call-ID: 3082416d51ef8b8a3c7283a00fb4cfff@216.239.128.31 CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- -- Forwarding SIP/user-b1e25258 to '9999' prevented. vidar*CLI> Disconnected from Asterisk server Executing last minute cleanups