similar to: AGI "RECORD FILE" for a video message

I am reading comments on the Wiki for meetme http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe from 2004 about how and AGI does work with non zap channels. Is this still valid 3 years later and 1.4.4? How do I bring people into a meetme and play a message to all of them when they are on SIP channels? Jerry

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)

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 =

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

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 =>

I am running asterisk 1.4.18 trying to use MeetMe and option b. I am getting permissions denied failed to execute conf-background.agi on the CLI lrwxrwxrwx 1 root root 37 Mar 17 10:11 conf-background.agi -> /home/silentm/bin/conf-background.agi my conf background is a symbolic link - then my permissions are : [root at devcentos5x64 src]# ls -l /home/silentm/bin/conf-background.agi

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 have setup as per the monitor example configuration on the wiki site and all works well for an extension dialing 8 then the number. However, if I dial from an AGI script the recording stops after a few seconds. I see an extra answer in the console and suspect that is the reason. Could any kind soul help me to get around this? Extensions.conf.. exten =>

Is there a way in asterisk to Activate/Clear the blinking light on polycom phones indicating VM. Either from an AGI or some way in the dialplan? I want to be able to control this light for from my application. I have an AGI to do something similiar to VM and want to light /clear the light myself. Thanks, Jerry

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

Hi All, I've been playing about with the RECORD FILE agi function and am finding two distinct problems with the resulting wav file when using a non zero sample offset. Specifically, I call the function with a zero offset and a given filename (the "original" recording), and then later call it with the same filename and a non-zero offset (the "overdub"). When I do this, I

Hi. I am using the 'get_data' function from an AGI, and i find that sometimes when users call in, it won't play the full gsm soundfile, and when i try to press a number (or pound, or star), nothing will happen - it just hangs there... anyone else experience this? - Dominic Son "It is not the force of a stroke that makes fine art" -------------- next part -------------- An

Hi, I have a question about this here: Asterisk-Version: 11.10.2 File: res/res_agi.c Line: 2377 [...] static int handle_recordfile(struct ast_channel *chan, AGI *agi, int argc, const char * const argv[]) 2304 { 2305 struct ast_filestream *fs; 2306 struct ast_frame *f; 2307 struct timeval start; 2308 long sample_offset = 0; 2309 int res = 0; 2310

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

I want to record audio in ADPCM format. According to the "show codecs" output of Asterisk, it looks like it supports adpcm. But I do not know what to tell the "RECORD FILE" directive in my AGI script. The RECORD FILE command usually has this form: RECORD FILE <filename> <format> <timeout> [BEEP] It records fine in WAV or GSM if I enter "wav" or

I have 3 phones P1 is a non video phone - grandstream P2 is a Grandstream GXV3000 P3 is a Grandstream GXV3000 Using P1 to place a call to P2 I get audio only (as expected). Then on P1 I transfer the call to P3 and I still only get audio. At this point shouldn't the two video phones P2 and P3 say to each other we are video and so startup the video stream??? This is not working at this time?

[apologies for no line wrap; config lines at bottom] I have mentioned on several threads here that the Monitor application doesn't do exactly what one would expect: the originating and answering legs of a call are unsynchronized by the duration of the interval that it takes for the answering leg to pick up the phone. This can be very distracting in a final mixed version of the file. Brian