search for: call_log

...ed off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing AMD("Local/91441425477394 at default-b9f2,1", "2000|2000|1000|5000|120|50|4|256") in new stack -- AMD: Local/91441425477394 at default-b9f2,1 0000000000 (null) (Fmt: 6...

...nter and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works perfectly. Normal Hang Up : ----------------------------- Quote: vici*CLI> -- Executing AGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in new stack -- Called VOIP74/17274507674 == Manager 'sendcron' logged off from 127.0.0.1 -- SIP/VOIP74-09ecb770...

...ew stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 -- Executing AGI("Local/192*168*000*002*78600051 at default-230b,2", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("Local/192*168*000*002*78600051 at default-230b,2", "SIP/192*168*000*002*78600051 at sip64||tTor") in new stack -- Called 192*168*000*002*78600051 at sip64 == Spawn extension...

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle_response...

...t; '19700101075955' limit 1;| == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Spawn extension (default, 8365, 3) exited non-zero on 'IAX2/u32218094-1' -- Executing DeadAGI("IAX2/u32218094-1", "call_log.agi|h") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi +++++ CALL LOG HUNGUP: |1139506471.2|IAX2/u32218094-1|h|2006-02-10 1:34:56|min: 0.37| -- AGI Script call_log.agi completed, returning 0 -- Hungup 'IAX2/u32218094-1' == Manager 'send...

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition? Thanks, Erick. --

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38 WARNING[2884]: chan_oss.c:585 setformat: Unable to re-open DSP devic...

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

...5060 Unmonitored 3210/3210 192.168.0.112 D 5060 Unmonitored 4 sip peers [4 online , 0 offline] -- Executing Ringing("SIP/3210-084eaa80", "") in new stack -- Executing AGI("SIP/3210-084eaa80", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/3210-084eaa80", "SIP/3213)|30|to") in new stack Feb 3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213) Feb 3 12:42:25 NOTICE[10368]: a...

...ditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2615) Verbosity is at least 21 -- Executing AGI("SIP/cc101-b7a07420", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7a07420", "SIP/17275691533 at sip8||tTor") in new stack -- Called 17275691533 at sip8 -- SIP/sip8-0825f9b0 is making progress passing it to SIP/cc101-b7a07420 --...

...e in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4. here are the CLI actual logs: -- Executing SetAccount("Local/50015308467418@default-ca2e,2", "XXXXXX") in new stack -- Executing AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi -- Zap/22-1 is proceeding passing it to Local/50015308467410@default-fb03,2 Mar 26 08:20:26 NOTICE[21367]: channel.c:2421 __ast_request_and_dial: Don't know what to do with control frame...

...s ringing -- Zap/33-1 is ringing -- Zap/34-1 is ringing -- Zap/35-1 is ringing -- SIP/operator1-e77f answered Zap/23-1 So how can I capture this value and write it to mysql? I already have this: $my.query("UPDATE call_log SET endtime = NOW() WHERE id = #{call_log_id}") And i needed to do something like: $my.query("UPDATE call_log SET endtime = NOW(),answeredby= #{$agi.WHOANSWEREDTHEPHONE} WHERE id = #{call_log_id}") And in above example...

...card showing OK with dahdi_tool, this NT mode. whenever I am dialing from server i am not able to connect the call . in Cli below mention warning is comming . please what is the mistake with me . help me Executing [0559566768 at default:1] AGI("Console/dsp", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing [0559566768 at default:2] Dial("Console/dsp", "Dahdi/g0/0559566768|55|tTo") in new stack [May 31 01:10:09] WARNING[30356]: channel.c:3443 ast_request: No channel type reg...

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

...stroying 1, having received hangup Sep 27 16:20:36 VERBOSE[10049] logger.c: == Spawn extension (default, 8600051, 1) exited non-zero on 'IAX2/172.22.0.5:4569-1' Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Executing DeadAGI("IAX2/172.22.0.5:4569-1", "agi://127.0.0.1:4577/call_log") in new stack Sep 27 16:20:36 VERBOSE[10049] logger.c: -- AGI Script agi:// 127.0.0.1:4577/call_log completed, returning 0 Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Executing DeadAGI("IAX2/172.22.0.5:4569-1", "agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI-----NODEBUG...

...back(transfer) ;play transfer (please hold whilst i try to conenct you) ;exten => s,3,Macro(record-start) ; WORKS but starts recording before connect ;exten => s,4,Dial(${ARG2},10,tT)) ; WORKS but starts recording before connect exten => s,3,SetVar(CALLFILENAME=/tmp/call_logs/${ARG1}/${TIMESTAMP}-${CALLERID}) exten => s,4,Dial(${ARG2},10,tTM(record-start)) exten => s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) ;--<SNIP>-- ;----------------------------------------------------------------------...

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

...Span 4" (MASTER) AMI/CCS RED span=4,4,0,ccs,ami # termtype: te bchan=10-11 dchan=12 echocanceller=mg2,10-11 # Global data loadzone = us defaultzone = us [root at go asterisk]# extensions.conf TRUNK=Dahdi/g0 [default] exten => _0XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _0XXXXXXXXX,2,MixMonitor(/var/spool/asterisk/astrec/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _0XXXXXXXXX,3,Dial(${TRUNK}/${EXTEN},,tTo) exten => _0XXXXXXXXX,4,Hangup exten => _XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_...

Help! I'm (still) having issues with Asterisk Queues. I want to implement a queue so that callers get the 'all our staff are busy at the moment, your call has been placed in a queue and will be answered by the first available operator. You may press 1 at any time to leave a voicemail' announcement, then they can press 1 and leave a voicemail. Documentation with * 1.2.x (I'm

...; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls, exten => _00034XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _00034XXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},55,To) exten => _00034XXXXXXXXX,3,Hangup ... ... and it works fine, but I need to start working with my second span & I don't know how to add it in extensions.conf file. -------------- next part -------------- An HTML attachment was...