similar to: VMS - AGI

Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From:

Hi all: Does somebody know how many meters of cable is supported between Asterisk Digium-FXO cards and the analog telephone ? Thanks, Angel

Hi all, Is there a way to get the "called id" (the B number) with AGI perl ? I know how to get the caller id which is working fine and is just below: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $callerid = $input{'callerid'}; $AGI->say_digits($callerid); } Thanks in advance, Angel.

Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? My asterisk version is ====>>> Asterisk CVS-HEAD-08/18/04-22:30:24 Thanks Angel. 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35

I ran in to an almost impossible ability to run MeetMe on our poweredge. Our PE is a 4 proc with OHCI and we had no need to buy any PCI cards other than for running MeetMe. If you know how to recompile your kernel, recompile it and make RTC a module (its compiled in by default). Then you can use zaprtc as your timing source. Be sure to `rmmod rtc` before you `insmod zaprtc` because zaprtc is a

Hello. I've just installed asterisk-1.4.21.2 zaptel-1.4.12.1 chan_ss7-1.0.10 libpri-1.4.7 I am using Sangoma A104 card with wanpipe-3.2.7.1 drivers. My OS: Ubuntu 8.04 Server Kernel: 2.6.24-16-server I am getting a choppy GSM playback and too many defunct AGI processes when channel closes. i am using Perl or PHP, also 'agi-test.agi' going to defunct too... I was able to playback GSM

Dear All, I seem to have stumbled across an AGI problem; I have written an AGI Script (bottom of this email); The script does the following; Makes a CDR entry when called Records the call Updates the CDR Finds a corresponding DNIS from the SMDR table (captured via a serial port logger) Matches up the record and updates the CDR. The script works perfectly in my test lab and has been doing so

A better explanation can be found here... http://www.digium.com/index.php?menu=faq#TDM%20&%20Analog_0 > -----Original Message----- > From: Benjamin on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: Monday, September 27, 2004 11:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FXO question

not able to run my php from AGIi am using asterisk 1.8.13 (debian)i am able to make call file using php command line..but when executing php from AGI, it is not working..kindly see the attachment if bellow text is not readable...___________________________________________________ File: /etc/asterisk/extensions.conf[call]exten => call,1,Answerexten => call,n,Playback(hello-world)exten =>

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

Hello- I've been writing a number of AGI scripts in Perl, and so far everything's working ok. However, yesterday I tried the AGI command "RECORD FILE" for the first time, and my channel locked up. Trying to stop asterisk produced a segmentation fault. There may be a bug here, but first let me make sure that I have the command line right (as the documentation is a bit sketchy)

Thanks Andy ! that is I was looking for. It works fine. Angel. >Date: Fri, 11 Jun 2004 00:55:04 +0200 >From: "Andy Powell" <andy@beagles-den.demon.co.uk> >To: asterisk-users@lists.digium.com >Subject: Re: [Asterisk-Users] How to get the Called id >with AGI >Reply-To: asterisk-users@lists.digium.com >On 10/06/2004 at 14:40 Angel Diaz wrote: >Hi all,

Apologies if this has been brought up before, but extensive googling and digging through my list archive didn't turn anything up. Basically, I'm working on an AGI web app and need to read some digit input. I'm having multiple issues with asterisk interpreting agi commands at the moment, but I figured I'd start with this one. when I call GET DATA or STREAM FILE I don't

Uuugh..for the life of me, i cannot delete sound files using "EXEC System(rm /var/lib/asterisk/sounds/blah.gsm)" through AGI the AGI debug log indicates the command executes successful ( equals 0) but my files are clearly still there. If i try System(rm ...) in my extensions.conf diaplan it'll work there. Is there a bug in the AGI to use "System" ? because i tried to

Hi, A quick and dirty (aka Rapid Application Developement) AGI script implement using bash shell. No need to invoke a 10MB perl engine to process simple asterisk agi scripts. I found it to be very useful in learning the AGI interface. For example, I learn that AGI won't execute the next command until you read the results from STDIN. Enjoy, Sunny Woo Solution Consultant Avantnix

I would appreciate any ideas of what I'm doing wrong on this. My dialplan calls an AGI which records a file. That works, but I'm trying to find a way to determine whether the caller pressed # to stop a recording before the maxtime expired, or if the recording ended due to reaching the max timeout. The $fx variable in the below agi excerpt always returns 0. $res =

Hi All. I have many days reading and research about asterisk and vicidial. I thing this issue is about asterisk and doesnt about vicidial. Isn't it? I have a problem with theses application (I already ask for help in vicidial forums), but I can not fix it. I have debian 5 with asterisk 1.2.24 and vicidial 2.0.4. This server has a IAX tunnel with another asterisk server B which connect to

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

I have used the AGI call "RECORD FILE" before for an only audio message. I pretty much always used "RECORD FILE filename gsm" before. What paramater do I use for gsm so that if video is present it will record video also and still work for only audio. Will this recorded file (with video) play on a video phone when calling out with a call file in outgoing spool? Thanks, I am