similar to: DeadAgi application not exiting

On Fri, 10 Jul 2009, Jared Mauch wrote: > I need the 'talking' information to better identify rogue people > on bridges. I'm a 1.2 Luddite so I don't have all these fancy new features :) A different solution to a similar problem. I had problems with abusive callers in my conferences. I whipped up some dialplan and AGI mojo to let an admin mute and unmute individual

Yes, I know, but I'm a 1.2 Luddite. 1.2 on 32 bits has been stable and rock solid for almost 10 years, but the client wanted to try it on a 64 bit version of CentOS 5.9. It all seems to be working OK, but when the customer service agents transfer a call they got from a queue, Asterisk crashes. Has anybody else had issues running 1.2 (or similarly ancient versions) on 64 bit OSes? Can

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

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

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

I have what my SBC/ATT representative calls a "Supertrunk," but he can't tell me the specifics I need to know to configure Asterisk to work with it. By fiddling about, I've come up with the "almost working" configuration below. This works except that it takes about 4 seconds from when the console says the line is answered until it plays my prompt. Watching the

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

Topic 6: RE:unable to execute command hi there i tried to execute the command as suggest like exten => 1987,1,Playback(posix-restarting) exten => 1987,2,wait(1) exten => 1987,3,System(/usr/bin/python /home/docas/Desktop/mess1.py) exten=> 1987,4,Hangup it still doesn't work,now it does it give unable to execute command but it doesn't reach the system command it just

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.

Wrong list. asterisk-dev is for changing the C source code of Asterisk. I don't think AGI's "count" or are considered for inclusion into the subversion repository as stated by one of your conditions for payment. On Wed, 29 Apr 2009, Alistair Cunningham wrote: > I'd like to offer a bounty for a feature for Asterisk where an AGI > program can park and retrieve calls

hi, all I am new to use DeadAgi, can anybody help me how to use DeadAgi, actually i want this, when caller hangup his/her phone, i want to send packet to my other app that check caller hung up done.

* The exit behavior of the AGI applications has changed. Previously, when a connection to an AGI server failed, the application would cause the channel to immediately stop dialplan execution and hangup. Now, the only time that the AGI applications will cause the channel to stop dialplan execution is when the channel itself requests hangup. The AGI applications now set an AGISTATUS

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 =>

system() to execute this script since it is (obviously) not really an AGI. I'm guessing that system() would be slightly more efficient than agi(). Both require a process creation, but agi() requires (slightly) more Asterisk resources in setting up the AGI environment. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards

Asterisk 11.11.0 on CentOS 6.5, installed from RPMs. OpenSIPS fronting Asterisk from a Tekelec T9000. I'm accumulating stuck channels. I'm googling now and I recognize that Friday afternoons are the worst time to ask questions, but I'm getting desperate because this is keeping me from rolling a system out to production. (Yup, I know. Who rolls out a system on a Friday

I'm writing some code that needs to access AMI in PHP. (I'll probably be doing AGI later as well.) I'm looking at phpagi (2.20), but it hasn't been updated since 2010 and appears to be a bit behind current Asterisk -- No event handler for event 'fullybooted'. What PHP framework/library are you using -- and why? -- Thanks in advance,

I'm trying to pass a variable between servers using "setvar" in iax.conf. I have a box (ts2) with a t100p in it. It answers the call and dials another box (ast0) via IAX. I want to pass a variable along with the call from ts2 to ast0. I'm running CVS-HEAD-03/07/05 on ts2 and ast0. ts2's iax.conf: [general] disallow = all allow

is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my

Wrong list. asterisk-dev is for changing the C source code of Asterisk. That's part of why you didn't get a response yesterday. On Wed, 29 Apr 2009, Juan Miguel Quiros Arrieta wrote: > I have to develop an application using the VeriFone vx510 device and I > read this device needed or could use a PPPoE connection in order to > validate and send all information collected from

Trying to redirect to -user... On Tue, 7 Jul 2009, Bryant Zimmerman wrote: > Hey guys I posted this earlier and did not get any responses. You posted what appear[s|ed] to be a user question to the dev list. I did reply (on June 3), but I may have mis-understood. > I am working on some AGI development that requires control of audio file > playback. The control stream file is working