similar to: Reading sound in eagi script and recognizing DTMF sounds at thesame time ?

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.

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

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

How can I execute an EAGI script with params from extensions.conf Example python script: InfMsg -s 1 in my extensions.conf exten => 492,1,Answer exten => 492,2,eagi,InfMsg -s 1 exten => 492,3,Hangup() It doesn?t work my * report... -- Executing [92@telpin-112:2] EAGI("Zap/4-1", "InfMsg -s 1") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/InfMsg

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

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

>>You say that each card generates an interrupt in every 1ms. >>So for 6 cards, each card needs to be processed atleast in >>166usec(1ms/6) in their interrupt_routine. Is it teoritically impossible on a >>perfect configuration/system? >In theory there isn't much difference between theory and practice. In >practice, there is. >In other words: you're

print to standard error output in your perl script: print STDERR "This is how perl-AGI prints to Asterisk CLI output\n"; MATT--- -----Original Message----- From: Robert Rozman [mailto:rozman@fri.uni-lj.si] Sent: Tuesday, August 24, 2004 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Perl AGI - no output from agi script to Asterisk Hi,

Hi all, I would like to do a very complicated system. Briefly what i want to do is; when a call came from SIP channel, I want to bridge it to another SIP account. This is normal part, but when call is over (either side hanged up or briefly say send BYE message), I don't hangup and echo a DTMF to one channel and hangup after that. How may do this with AGI or EAGI. Yours -------------- next

With our current setup, we have an older avaya system which is linked with our asterisk system via a em wink connection. When you press "2" on the avaya network, it will jump to our asterisk box and then sends DTMF digits. Asterisk listens for those numbers and then responses as soon as it has a match. The problem is with having a "send to voicemail" option. Right now, a user

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

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

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

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

I'm trying to set up a system so that I can record a conversation over SIP. Monitor and the like don't work so well for me, because I need to pipe the conversation to other programs in realtime, rather than record to a file, so I've been trying to use EAGI instead. (if anyone has any other suggestions about this, it would be greatly appreciated!) At this point, I'm a little

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

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

On Thu, 2003-12-18 at 15:40, Kevin Bockman wrote: >> 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?

I have an AGI script written in PHP that worked great with Asterisk 13. I'm porting it to an Asterisk 20 site and have a strange problem. I tried running the script from the command line and it works fine; I see the script commands written to stdout like VERBOSE "SmartScreen v1" But when run from asterisk the CLI shows: [2023-06-30 15:50:47]