similar to: [Asterisk-Dev] C++ AGI debuggin

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

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thanks Billy. I replaced print "STREAM FILE $filename \"\"\n"; with print "EXEC PLAYBACK $filename \n"; and it worked fine. Interestingly when I did print "STREAM FILE beep \"\"\n"; within the script, it worked. If I wasnt a newbie to asterisk I wouldve thought this to be strange. >From: "William Piper"

Hello, Does anyone know any solution to this? Or is Asterisk-Dev a more suitable list to ask this question? Best regards, Vlasis Hatzistavrou. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Vlasis Hatzistavrou Sent: Thursday, February 16, 2006 3:43 PM To: asterisk-users@lists.digium.com Cc: 'Vlasis

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

Actually, there was recently a bug fixed regarding multipart SDP parsing in chan_sip. That should have fixed the issue with CS1000s and SIP (among other things). I haven't actually tried it yet on my CS1000, but it should work. Regards, - Brad ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Jerry Geis Sent: Tue 12/26/2006 3:51 PM To:

But for us? _____ From: William Boehlke [mailto:william.boehlke@signate.com] Sent: Wednesday, January 11, 2006 2:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Second edition of my * book has been released $39.95 retail. _____ From: asterisk-users-bounces@lists.digium.com

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On Oct 22, 2014, at 10:33 AM, Joshua Colp <jcolp at digium.com> wrote: > Paul Albrecht wrote: >> Really? Shouldn?t something this major affecting the entire Asterisk >> community get discussed on the lists? Any idea what Leif is talking >> about when he says the community is in transition, moving from dial >> plan model to external control. > > It was

It's a udev configuration. Read UDEV.txt, or, read the AMP wiki http://aussievoip.com for a step-by-step. --Rob -----Original Message----- From: asterisk-users-bounces@lists.digium.com on behalf of Jachin Rupe Sent: Wed 7/09/2005 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] /dev/zap* is not showing up (gentoo, portage,asterisk 1.0.8 hi

(This subthread is more appropriate to -users than to -dev, so it is crossposted only to mark its transition. Please reply on the -user list only.) What are the cheapest prices for (humans) transcribing voicemail to text as a service? The absolute cheapest, regardless of (known) quality - the quality only has to compete with (cheaper) automated transcription, which is abysmal quality. On Wed,

Hi everybody on the sambalist my server is running pretty ok now, but after examening my logs im trying to get rid of the last messages. 1) Tons of messages like this, ( i have 20 printers overhere ) log.smbd [2005/11/23 10:24:09, 0] printing/print_cups.c:cups_queue_get(790) Unable to get jobs for ipp://localhost/printers//usr/bin/lpq -P'pdfprinter' - printing is setup RAW,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi there, I tried the change from cyrus to dovecot. This was just some days ago and I'm really happy. Just a little problem. For testing I switched on some login debugging. Now I can't switch it off. Dovecot seems to ignore the "no" in the config file. <snip> auth_verbose = no auth_debug = no </snip> That is what I get

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Questions: Does anyone have a really STABLE asterisk system running about one year without need to restart the service or the SERVER ? Does anyone have a production Call Centre saled that don't lockup and is stable for 6 months ? I'm asking this questions because we have choose Asterisk for our call centre solution but, since the bugtracker only grows and people still want to stuck more