similar to: Execute EAGI script with params from extensions.conf

Hi, using EAGI variables like agi_request agi_channel agi_language agi_type agi_uniqueid agi_callerid agi_dnid agi_rdnis agi_context agi_extension agi_priority agi_enhanced agi_accountcode I get a lot of data about a call, but I need to obtain P-Asserted-Identity value from a SIP call. Are tehe any eagi variable to get that? Or have you any solution?? Thanks!!! -------------- next part

Hi, I'm trying to use Swift tts from eagi, my problem is when I send EXEC SWIFT <*prosody rate*=\'.8\' >Hello World\, this is a test\,</*prosody* >|0|1 Would I use a scape character? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100301/7132be4c/attachment.htm

Hello, I made a post to the forums (http://forums.digium.com/viewtopic.php?f=1&t=72901&sid=3d5c2717ca5ab7ad676957ae436d4b51) but haven't received any replies, so thought I'd try here. On my debian machine running asterisk 1:1.4.21.2~dfsg-3, I've been noticing that there's a problem with conferences (using both meetme and app_conference) and the audio sent out to an

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like also to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hi, we've connected Sphinx4 through eagi script (modified eagi example) to Asterisk. Users can now say their wishes - but for gradual evolution we would like to provide "older" way of DTMF navigation too - can we recognize DTMF while reading sound in eagi ? Any advice or examples ? Thanks in advance, regards, Rob.

Hi! If the caller hungs up while an eagi script is running, I can?t regiter the cdr manually at the end of the script. I tryied to trap SIGHUP but it didn?t work. I want to register my own cdr into the script because I have a lot of data that I need to put in the cdr. The 'h' option or DeadAgi aren?t a solution for me. Thanks -------------- next part -------------- An HTML attachment was

When I use an eagi script when play a message appear a lot of warning messages, but it play very well I?m using Asterisk 1.4.32 dahdi-linux-2.3.0.1 chan_ss7-1.4.1 Any ideas?? -- Playing 'ser002' (escape_digits=0123456789*#) (sample_offset 0) [Jun 11 18:12:45] WARNING[15807]: file.c:1300 waitstream_core: write() failed: Broken pipe [Jun 11 18:12:45] WARNING[15807]: file.c:1300

Hi. I just started trying to play with Sphinx. I followed their site as far as running sphinx-server. It is listening on the default port. I copied sphinx2-simple to another file and changed sphinx2-continuous to sphinx2-server. So, I ran eagi-sphinx-test under asterisk. What exactly is it supposted to do? Here's what I get: debian:~# sphinx2-simple2 sphinx2-simple: Demo CMU Sphinx2

Hello folks, I'm interested in detecting tones on specific frequencies with specific timing; for example, I'd like Asterisk to dial out and when the channel starts/call connects, listen for a 1200Hz tone that plays for 100ms. Is this doable with Asterisk using something already extant? After looking through documentation, mailing lists, and some of the source I had the idea that I might

I'm having a problem with conferences both meetme and app_conference, though I've done most of the testing with meetme. Essentially, I get little gaps in the audio - usually fewer than a dozen or so samples, though it does vary. They seem to occur at random, but I usually get one ever few seconds, on average. They also seem to delay some buffer somewhere, so that if I start recording

I've got a single phone line coming into an X100P. In extensions.conf I've got this: [inboundzap] exten => s,1,Answer exten => s,2,EAgi,hanguptest.agi I see the ring come in and Asterisk detects it and tries to do something with it: NOTICE[20492]: File chan_zap.c, Line 4017 (ss_thread): Got event 2 (Ring/Answered)... -- Executing Answer("Zap/1-1", "") in

Hi, need a quick help ... it should be easy but ... exten =>_9898,1,Answer exten =>_9898,2,VoiceMailMain(${CALLERID}@domain) Accepting overlap call from '342' to '9' on channel 0/2, span 3 -- Executing Answer("Zap/8-1", "") in new stack -- Executing VoiceMailMain("Zap/8-1", "@domain") in new stack As you can see there

Hello there, I'm trying to figure out how to run a PHP script on a remote machine and still have access to the audio stream associated with the call. Ideally, I'd love to play/record audio files directly from/to the remote server without having to copy them back and forth to the Asterisk server. What is the best way to do this? Is it possible to combine EAGI with FastAGI in PHP?

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

I've been trying to use the AGI get_data function for some time now, and can't get it to work. Today I reinstalled a clean system with Red Hat 8.0 (I had been using RH9, but was told * had problems with RH9) and downloaded the latest Asterisk CVS to install. I then downloaded and installed perl-asterisk-0.08. I have extension 502 pointed at EAGI(agi-test.agi). When I call that

The AGI debug command worked well, and I found the offending command: SetCallerPres(allowed) That worked in Asterisk 13, but from my google searching it looks like this command has disappeared in Asterisk 20 (actually everything after ver 13). I thought it was replaced with CALLERPRES(allowed) but this generated an error too in Asterisk 20. Is there a replacement command? -----Original

Hello, Question for developers: what is the asterisk way to integrate with ASR (speech recognition)? Question to the community: has someone done anything in this direction? On the first glance that shouldn't be too hard. One part is delivering audio to the engine (for example, main ASR players Nuance and Speechworks will be happy with RTP) - can be done via RTP forking. The other part is

Does anyone configure that or is that possible ? Thanks in advance -- Best regards Vlad

Hi I am currently developing a IVR application using PHP/AGI. I am using the PHPAGI class at http://phpagi.sourceforge.net/ to handle the commuication with my *. The application basically asks a caller to enter in some information which is then processed and a answer is read back out to them. I want the application to loop back to the beginning after giving the answer so they can try another

Hello! I have an asterisk setup where several sip devices are connected to an asterisk box. I am looking for a method that lets me know whether any of the sip devices is on hook / off hook / busy etc. I have tried the AGI command CHANNEL STATUS <channel name> but it returns with a message 'There is no channel that matches <channel name>' In concrete terms, my channel is