similar to: issue recording files in wav49 from AGI

Hi All, When I call the stream_file function all goes well if the user doesn't clear the call. But if I do clear the call (on the handset for example), I get the following exception: -- Channel 0/31, span 1 got hangup RESULT_LINE: 200 result=-1 endpos=28000 == Spawn extension (default, 600006, 1) exited non-zero on 'Zap/31-1'

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,

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

When I pin is being matched to ASTCC database I get this message and the call is dropeed. Oct 2 15:22:20 WARNING[327699]: file.c:475 ast_openstream: File your does not exist in any format Oct 2 15:22:20 WARNING[327699]: res_agi.c:435 handle_streamfile: Unable to open your == Spawn extension (from-sip, 77, 2) exited non-zero on 'SIP/2000-4d5b'

Hi Here is another patch regarding time in ogg123 which is more controvercial than the other one I send some days ago. (see attachment) When working with Daisy/SMIL [1] files it would be helpfull if the user could stop play at a specific time. Currently .ogg files are not allowed in the Daisy format but that will hopefully happen one day if I work hard at it. A clip in a Daisy file could look

Greetings Friends, I would be grateful if you can help me undestand how to make my R code more efficiently. I have read in R intoductory tutorial that a for loop is not used so ofter (and is not maybe not that efficient) compared to other languages. So I am trying to build understanding how to get the equivalent of a for loop using more R-oriented thinking. If I got it right one way to do that

Hi, I''m trying to use highlight ferret method with trunk aaf and 0.10.1 ferret. In my search display I use: Myindexedclass.ferret_index.searcher.highlight(@query, result_line.id, :content) * searcher is a protected method; how can I access to the searcher from aaf ? * is the doc id in aaf the same as my model id ? * is the first param, query, the string query or the query object ?

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

Hi Guys, I have already tried this one on the developers list. I have not been successful getting much back there and I have notified them that I will post this on the users list instead. Hopefully somebody have tried something similar and can help out. I am developing AGI scripts on Asterisk and have run into some very strange behaviour and I think this is a bug, but I am not completely sure.

My users dial *120 get to an IVR menu that plays their balance and then ask them for a voucher. Ater the balance is played and the request for the voucher is played the user don't hear any other audio from the asterisk box. I can see the asterisk server playing the files to ask for the voucher again but the user cannot hear any thing. Has any one seens this issue with IVRs. I notice a

Hi all, I have a very simple AGI program in Java, but it "sometimes" hangup before the whole process is completed. Looking at the /var/log/asterisk/full, I found the following: AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Rx << STREAM FILE beep "" Jan 20 07:32:30 WARNING[-169026640]: Wait failed (No such file or directory) Jan

-- jand. more than just a group Asterisk AGI Command GET DATA is usually of this form GET DATA timeout max_digits When I execute this command, I get only a single digit, regardless of what the value of max_digits is, Also the script quits Immediately after the press of the digit regardless of what the value of timeout is, This is really un-desirable as I will like to GET multiple DTMF digits

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 <<

I have a "support call" AGI script that has been working flawlessly for a couple of years now. It dumps the customer into a MeetMe conference room, then dials a bunch of support engineers, and connects anyone who accepts the call into the conference room. The conference room is recorded. After the support call is over, the recording is emailed to a list for quality control, etc. It

Hi, Happy Thanksgiving to those of us in the USA... Been trying to debug an AGI (in C) on 1.4.26.2. I blind transfer a call to this snippet of dialplan: exten => 999900,1,DeadAGI(pq.agi,999950) pq.agi then plays a prompt (which I hear just fine): [Nov 26 02:42:47] VERBOSE[28721] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/pq.agi [Nov 26 02:42:47] VERBOSE[28721]

I have this in nodes.pp $puppetmaster_fqdn = generate("/usr/bin/facter","fqdn") and this in a template http://<%= puppetmaster_fqdn %>:8080 When puppet runs, this is the result: http://puppet.home :8080 Anybody any clue to whats causing this? I''ve tried -%> -- You received this message because you are subscribed to the Google Groups "Puppet

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 With these one can build 2.4-beta (AKA 2.3.99.0) with mingw32 (gcc-4.7.0). Note, however, that win32 subdirectory is: 1) Lacking a complete Makefile.am, so it can't be built with autotools. 2) Depends on MFC, and MFC is not provided by any MinGW toolset (and MS' MFC is, most likely, highly incompatible with MinGW). So you only get a console

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

The special wav49 ms hack doesn't produce correct WAV files on output. they cannot be played back on windows media player nor winamp. for example a typcial voicemail message with 9 seconds stops after 6 seconds playback with the error: "invalid fileformat (Error=8004022F)" attached is such a file. in the meantime I use the uncompressed wav format and postprocess this in

Can anyone point me in the direction of a utility which will let me determine the length (in seconds) of a wav49 file created by Asterisk? Many thanks, Malcolm