similar to: Subject Change: Playback/background App over Network (was: AGI STREAM FILE not working?)

Hello, I have an asterisk 1.2.18 working fine, the only problem is that all applications that play audio, sound like "tremolo" or "vibrato", but musiconhold plays fine. The same audio file (wav, mp3, ...) works fine with Musiconhold() but not with Playback() or Background()... If I move app_playback.so from this system to another asterisk, playback works fine... Do you

Hello, using 1.8.4. using a very simple local AGI script in bash which has only one line in it: echo -e 'STREAM FILE welcome 123 \n' dialplan: exten => 5150,1,Answer() same => n,Set(CHANNEL(language)=en_AU) same => n,AGI(testagi.sh) same => n,Hangup console output: -- Executing [5150 at AllPhones:1] Answer("SIP/PBX-00000024", "") in new stack

For some applications, storing recorded audio (prompts and caller recordings) as a BLOB in MySQL has advantages. So, once I have the audio in the database, how can I play it? Creating temporary files seems so tacky. Is there another way to playback or background audio either by specifying a URL or from a memory buffer (either C or PHP)? -- Thanks in advance,

In the following deployment of web app on ONE pc-based server,running linux,which can support the most amount of concurrent users as well as the shortest latency? 1 apache+mongrel 2 apache+mongrel_cluster 3 lighthttpd+mongrel 4 lighthttpd+mongrel_cluster 5 lighthttpd+fastcgi Thanks charlie -- Posted via http://www.ruby-forum.com/.

I am having a problem with "record_file" working properly depending on when it is called -- basically if it is called immediately upon a call, it acts like it does not hear anything from the callers phone (yes, my phone is setup properly and functions fine otherwise)... if I do a "background" or "festival" command before calling it, it works fine. Details below:

I have a pretty basic Mandrake 10.2 w/KDE 3.2 and I installed Asterisk-1.0.1-2mdk. I installed the source of main and contrib from ftp, so at the install time I accepted all the packages needed to be installed too. The installation went smooth, but when I try to execute asterisk (#asterisk -vvv) I encounter few warnings I end with an error. At this point I didn't touch any conf file, I was

Hi all, I am using asterisk-1.6.2.10. I changed say.conf script for customized number reading. In the extension.conf: -------------------------- [number-to-voice] exten => 8765,1,playback(num:344345,say) exten => 8765,n,hangup It executes corresponding say.conf script and produces good results for me. but when I write it in agi does not working. Here is agi debug output from asterisk.

Hi all, I've read a bit about fastcgi, and wondered if it might be an interesting solution for e.g. running php in a user-switched environment or running ruby-on-rails etc. But I wonder why nobody (searched on Google) seems to use CentOS with mod_fastcgi for Apache 2.x. And I also wonder why people actually using Apache 2.x and ruby-on-rails with CentOS did run a lighthttpd-fastcgi

Hi, On the api doc it is written that if lighttpd is installed, rails will use it as its web server instead of WebBrick. I have installed lighthttpd and verified it works (for example on the default port 80) but when lauching script/server, it still uses WebBrick. I wondering if I need to edit /etc/lighttpd/lighthttpd.conf myself or configure something else. Any idea? Thanks Oscar

I have the following python AGI script. I know it's been abstracted, but it's still pretty easy to see what's happening. self.agi.channelAnswer() self.agi.wait(1) self.agi.execCmd("background","enter-conf-call-number","") self.agi.execCmd("Read","confNum|||","") confNum =

Trying to redirect to -user... On Tue, 7 Jul 2009, Bryant Zimmerman wrote: > Hey guys I posted this earlier and did not get any responses. You posted what appear[s|ed] to be a user question to the dev list. I did reply (on June 3), but I may have mis-understood. > I am working on some AGI development that requires control of audio file > playback. The control stream file is working

Ok, now I'm really getting places. Something else has got me stumped - asterisk does not receive any callerid information. If I set up an extension as follows: exten => 444444,1,Sayalpha(${CALLERID}) exten => 444444,2,Hangup I get the application saying Executing SayAlpha("Zap/1-1","") in new stack which implies that the ${CALLERID} variable is blank. Do I need

strace on file access in asterisk shows that * is not even attempting to access the voice files. If I *manually* load app_playback.so, app_macro.so, and then pbx_config.so, I they will load and I get a dialplan. Ok, that's a problem -- autoconf is clearly not working, or there's some other related issue. So I try to use the demo and do "dial 500". This should connect and

This is the core trace (gdb) bt #0 0xb7e5a231 in strcasecmp () from /lib/libc.so.6 #1 0xb7ce0a3f in local_ast_moh_start (chan=0x82496a8, mclass=0xb720f828 "default", interpclass=0x0) at res_musiconhold.c:646 #2 0x08083695 in ast_moh_start (chan=0x64, mclass=0x64 <Address 0x64 out of bounds>, interpclass=0x88 <Address 0x88 out of bounds>) at channel.c:4614 #3

strace does show that that modules.conf loads: > open("/etc/asterisk/modules.conf", O_RDONLY) = 8 And that I do get some of the channels loading, e.g., the modem channel: > open("/usr/local/lib/asterisk/modules/chan_modem.so", O_RDONLY) = 8 And if I load the apps via "load app_playback.so", >

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

Hi everyone, I'm having an odd issue. I've been doing some testing over the past couple weeks on some Asterisk modules / utilities, but have bumped into a problem which I can't seem to resolve. Asterisk can't seem to play the default sound files (ULAW) in my environment. All necessary debugging information is included below. I'd love to get anyone else's thoughts on this,

Hi, What is the best way for you to simulate a provider behavior in modules ? For exemple if you have an webserver module and want to have a apache2, lighthttpd prividers. How would you do so the call to the modules functions stay like: webserver::addvhost { ... } and not webserver::apache2::addvhost Any best practices you can share ? I know that conditional include would not work

On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] <all.eforums at gmail.com> wrote: > Hello All, > > Probably a silly question, but we're wondering if people have had any > experience and have data to demonstrate if the performance of the Asterisk > system might suffer in terms of latency etc. if we're to have it retrieve > sound files from a database using odbc as

i call to my tollfree number buy my CLI send the next error: Aug 14 08:15:22 WARNING[25931]: format_wav.c:169 check_header: Unexpected freqency 22050 Aug 14 08:15:22 WARNING[25931]: file.c:441 ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/procall3.wav Aug 14 08:15:22 WARNING[25931]: file.c:828 ast_streamfile: Unable to open procall3 (format ulaw): No such file or directory Aug