similar to: DeadAgi

Hi I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Please help. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070722/09df42e9/attachment.htm

Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100127/b13d6a62/attachment.htm

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It

Hello, I need to execute an agi in php. I have that: == Using SIP RTP CoS mark 5 -- Executing [0170725000 at mnupprx1:1] Answer("SIP/33179977999-b6c18478", "") in new stack -- Executing [0170725000 at mnupprx1:2] GotoIf("SIP/33179977999-b6c18478", "0?6:3)") in new stack -- Goto (mnupprx1,0170725000,3) -- Executing

How can extensions.conf be changed to work with both Asterisk 11 and 1.4.X such that 1.4.X calls deadagi and 11 just calls agi as deadagi is no more. Thanks, jerry

Here is another log from the * server CLI, I reall hope some one can help me out on this one. thanks |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time < "" and lead_id != '';| -- VDAD get agent: |0|update of vla table: |127.0.0.1 |UPDATE vicidial_live_agents set

hi, all i want to create cd-rom with asterisk. how it possible. when i put disk in cdrom it boot automatifcally and auto-start installation like TRIXBOX. any idea. thnks, Bhrugu Mehta

Hi All; Really I need to know why when using the "h" in the exten =>, then we use DeaAGI with it? I am using vicidial and I see this line alot, so I need to know how it work (when it will be executed): exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME}) The question is: When

Dears; In asterisk 1.8, it is not more possible to use DeadAGI? Also, I found the below commands in the a2billing and I would to ask why it set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How? [a2billing-callingcard] exten => _X.,1,NoOp(A2Billing Start) exten => _X.,n,Answer() exten => _X.,n,Wait(2) exten => _X.,n,DeadAgi(a2billing.php,1) exten =>

hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run " show applications" at cli> . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta

I downloaded the astcc calling card program. Thanks, it is very easy to setup and works Excellent. Anyway, it says to use DeadAGI to run it rather than AGI. I don't know what I am doing wrong. I just updated my asterisk from cvs and rebuilt and reinstalled. I do not have an application called DeadAGI. I have searched the source, google, etc. but have not been able to find anything.

Hi Everyone, Trying to run a php script after DeadAGI for A2Billing does it's magic. This is the dialplan: [a2billing] exten => _X.,1,System(php pre-call.php ${CALLERID(num)} ${EXTEN} ${UNIQUEID}) exten => _X.,n,AGI(a2billing.php,1) exten => _X.,n,Hangup() *exten => h,1,Wait(5)* *exten => h,n,System(php post-call.php ${CALLERID(num)} ${UNIQUEID})* As you can see above, I even

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

hi, all I am new to zaptel programming. can anybody help me how to start this. or any ref. site or matirial availabel. i want to use c lang. for this. thnks in advance. Bhrugu Mehta

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in

hi, all is there any way to send image on sip channel ? Regards, -- Bhrugu Mehta Sr. S/W Engineer (D&D) VOIP,Telephony Team India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100317/3cb8322a/attachment.htm

Hello, I want to create php rate script and I'm using Deadagi. But I allways get billsec 0 , or nothing. Can you help me to solve this problem... My extension.conf: exten => _123,1,DeadAgi(rate.php) exten => _123,2,hangup And my simple test php script rate.php #!/usr/local/bin/php -q <?php include_once (dirname(__FILE__)."/phpagi.php"); $AGI = new AGI();

Hi * users, We're using calls to external scripts through AGI at various points throughout our IVR system. We use these scripts to log certain events and to make certain choices that I wasn't sure would be possible in the dial plan. The problem comes with with the final call to our script. We use this line: exten => h,1,deadagi(log.php|{$service}|Hung up|${UNIQUEID}) I know there are

I get a warning every time I run DeadAGI from the h extension: -- Executing [h at CC:2] DeadAGI("Zap/7-1", "graba_dialer.agi") in new stack [Mar 11 20:50:10] WARNING[8598]: res_agi.c:2203 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI I use the agi script to do clean up and move the recording of the call. The only way the h

Hi all, The asterisk book states the following for using ODBC to connect to an MS database. ? The pooling and limit options are quite useful for MS SQL Server and Sybase databases. These permit you to establish multiple connections (up to limit connections) to a database while ensuring that each connection has only one statement executing at once (this is due to a limitation in the protocol