similar to: AGI Assistance

I am trying to figure this out... I'm sure it's simple, but I can't think of it right now.... In my AGI Script I am doing this... (This is done in Perl) $AGI->exec('Record', "/usr/local/apache/htdocs/demo/sound/$EmpNum%d:wav"); And after this is done.. I want to get the name of the file it created so I can store it in a database. Any thoughts.... Thanks, -gcc

Let me lay it out for you.... Call comes in over a T1 - Signal is em_w. The extension is seen as *<callerid>*<last 4 digits of number being called>*. Which is fine in it self. I have my extension.conf file set up as follows... [did] ; Receive call as *<calling>*<called> exten => _.,1,Answer exten => _.,2,Cut(CALLING=EXTEN,*,2) exten =>

Hello, I am having some trouble with the Asterisk::AGI perl library. It seems that the AGI->Exec() command is causing me a problem. Here's the line in my AGI code: $AGI->exec('Record',"$vmfile:wav, 30"); I'm trying to record voicemail to the file name stored in $vmfile with a silence timeout of 30. However, this is not being parse by AGI or Asterisk correctly,

I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which plays few wav files depending on external parameters. Can I have a dial plan inside my AGI? If not, how do I accomodate user who needs to reach extension 2 from my agi? I have tried stream file and get data but the two commands did not work at all.

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 =>

I have the following code. When I call the extension, it either ignores the first "Hello there everyone", or says "hello" and moves on sometime stoping before it finishes hello. The rest of the text reads fine. Anyone else have this issue?? Thanks! require('/var/lib/asterisk/agi-bin/phpagi.php'); $agi = new AGI(); $agi->answer();

I have the below script that doesn't seem to be working. I don't know if I have something in the script wrong that I am just missing. Or if I don't have the php.ini set correctly for emailing This is the CLI output -- Executing [4099XXXXXX at port3_real:1] Goto("DAHDI/50-1", "newhire,s,1") in new stack -- Goto (newhire,s,1) -- Executing [s at

Hi, I am developing a call recording application on Asterisk 11.2 and have this configuration in my dialplan: [macro-ccdev2-rec] exten => s,1,MixMonitor(${ARG1},b) [outgoing-originate] exten => _X.,1,NoOp(Will send call to ${EXTEN}) exten => _X.,n,Dial(SIP/${EXTEN}@x.y.z) [outgoing-originate-rec] exten => h,1,Agi(agi://localhost/ajpbx.agi?path=uploadrec&callid=${CC_CALLID})

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

All; I wrote a PERL AGI script that prompts a caller to leave a message using print "RECORD FILE $recordfile wav # 60000 BEEP s=3\n"; When the caller is done, they need to press the # key. The message is then delivered. However, the message is not delivered if the caller simply hangs up when finished. If the user hangs up, the script ends right then. How do I keep on processing the

I just got cepstal working fine in the dial plan using code like: exten => 511,5,AGI(cepstral.pl|Welcome to my house finder. At the beep enter your zip code.) The php script it calls is based on the nerdvittles weather one so it calls a webpage which prints to the screen, the nerdvittles code uses system to generate the .wav file then has the dial plan call it via: //php script $retcode2 =

How can I use the application Monitor() in the Python AGI skripts? Thanks a lot. best regards, Feilx -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110201/c1256374/attachment.htm>

I'm expirimenting with Cepstral via swift.agi. It loads w/ no problem, but there is a fairly long separation between words -- almost as if it is processing one word at a time, rather than stringing a phrase together. When I run cepstral with a similar script, but not through an actual call from asterisk the separation between words is short and natural. I'd really appreciate

Hi Everyone, I'm wondering if anyone here write AGI's in compiled binaries. I'm writing a small Cepstral AGI in Freepascal/Lazarus. I know there are some other AGI's out there, but I wanted to add some more functionality than what is available such as having the AGI determine if the "data" argument is plain text or a path to a text file and act accordingly. The

Hi all I got a vps setup amazingly fast using the deprec recipes on Ubuntu edgy, but now I''m having a minor issue where my site loads for the main domain when I only want it accessible from the subdomain. This is a pretty typical setup, apache 2.2.3 -> mongrel_proxy_balancer -> mongrel cluster. I''m sure I''m missing something obvious in the config... I have the

A customer contacted us today concerning getting a VoIP to PSTN system with a few IP Phones setup. Asterisk should fit his needs. It is not a big job, but I think that this customer is going to need onsite work. Please contact me off list if you are an interested reseller in the Washington, DC area. Sean _______________________________________________ Sean Robertson NETXUSA p. 800-289-6389

Hello, I have written an AGI script for asterisk that uses google translate for text to speech synthesis. It supports a variety of different languages, local caching for the voice data and wideband audio. The voice in most languages is female and the quality of the synthesized speech is very high. More info about the script can be found here: http://zaf.github.com/asterisk-googletts/ the first

I'm a very new convert to Linux and have just installed Ubuntu 6.1/ Gnome. Because I need to run a couple of Windows apps, I also installed Wine and IE. Using winefile, I tried to install Front Page, but although everything looked as if it was going OK, it failed at the point where it was attempting to write Registry Values. The following screen opened on my desktop, which seems to be a log of

Is anyone successfully using the AGI script calleridnamelookup.agi (or anything similar) ? I get both name and number caller ID from my POTS line, but I'd save money if I had them deliver ANI only. I've downloaded and installed the AGI script calleridnamelookup.agi, but I always get -- Executing AGI("SIP/9525485560-5359", "calleridnamelookup.agi") in new stack

For reasons best known to myself, I call a python agi (PYST2 - love it!) which streams a series of very short files in quick succession. Like this: escape_digits = str("0") agi.stream_file(promptFile,escape_digits) and this is what I see on the AGI debug: <Local/s at root-00000061;2>AGI Tx >> 200 result=0 endpos=6784 <Local/s at root-00000061;2>AGI Rx <<