similar to: Can't get format_mp3 to work for music on hold

You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume > -----Original Message----- > From: Stefan Gofferje

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

Checkout http://www.voip-info.org/wiki-Asterisk+variables I believe that should have the answer for you. furthermore assuming that your number is always going to be 12 digits. exten => _NXX.,1,SetVar(mynumber=${EXTEN:0:12}) - will give you your number. Hope this helps. Umar On Sun, 13 Mar 2005 09:25:11 +0100, Harald Milz <hm@seneca.muc.de> wrote: > Hi, > > this

Hi all, I am trying to write an application to monitor queues using the Asterisk Management API. So far I have had some level of sucess, basically reverse engineering the protocol and the event messages using ethereal etc. I know there are a couple of pages on the Wiki that attempt (no dis-respect to who ever did it as it has been a great help) to document the API and was wondering if there

Does anybody know why I can NOT hear the MusicOnHold - using SJphone on another PC in our network (normal playback is not a problem) . See the * output and the line configured in extension.conf below (also mp3player does not function) Any suggestions? *Asterisk output:* *CLI> -- Executing WaitMusicOnHold("SIP/jeroen-bf54", "30") in new stack --

Hi All, I hope someone can help. I have a system that I have recently upgraded to latest CVS and my voicemail is not working from mysql database. I get an error on the console saying " No entry in voicemail config file for 'number'" whilst there is an entry in the database for the specified number. It seems like app_voicemail is no longer checking the database even though

Hi I am trying to use SetCDRUserField in an agi script but with no success. I am using the CDR mysql addon, however I can't see it being at fault as my attempt is not doing anything to the CVS CD either. has anyone used this, any hints guidence would be greatly appreciated. The syntax I am using is like so .. res=DoExec('SetCDRUserField','12345'); and then dialing the

Hi, Has anyone manage to comile app_dbodbc or ast_data with the latest stable release (1.09). If so can you give some guidence on howto do it as I have trouble getting either working. Umar

Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]

Hi, i have a problem with iax2 and ringtone. Here is the call path pstn -> asterisk -> iax -> firefly or any iax phone. My problem is when i receive a call on my iax phone, the ring sound is very distort and bad. If i open my sip phone, and receive a call from my pstn, the ring is like dring dring, very normal. Otherwise, it is like a machine gun with iax Help would be really

Hi, Has anyone done any AGI scripting in pascal. I would appreciate help anyone can offer. My understandin on AGI scripting is very flaky, I am assuming whatever language is used the application needs to be compile and made executable. So if I write a script in pascal, I would compile it with something like freepascal and make it executable. Thanks Umar Sear

I'm trying to use Asterisk for its voicemail capabilities while interfacing with a legacy Toshiba PBX. Is there a way to have Asterisk send a DTMF code to an extension to turn on the message waiting indicator light? When a user leaves a voicemail, I want Asterisk to pick up one of the lines attached to it, and then dial #63<ext>, which is what sets the message waiting indicator light

Hi List, Forgive me if this has already been covered. I did go through past messages but could not find anything. I want to setup a queue like scenario where users don't need to login/logout. Basically I want to define a list of extensions that will be rung when a call comes in. The sequence in which the extension are rung needs to be intelligent, in the way queues are. For example, it

Dear list, I have been using asterisk for some time now. However I have never used it with any of the digium or compatable cards (Purely used for SIP). I understand that for using Meetme, I need to have a timing device, which could either be hardware or zrdummy etc (I am not using any right now). Can someone tell me if the timing device is needed for voicemail and other applications too?. I am

Hi, I have compiled and installed app_prepaid module. But have problem when connect to postgres database. I guess so because after key in card number, it always play prepaid-no-aaa voice file. Anyone succeeded in configuring the app_prepaid for prepaid calling service for asterisk? Please help. Ps: where can I view the log file for this module. Thanks. Tom --------------

; ; Music on hold class definitions ; [classes] default => /var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => quietmp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters (specifically embedded spaces) ;manual =>

> icc generates testq for 0-30 and btq for 31-63. > That seems like a small bug in the bit 31 case. You can’t use testq for bit 31, because the immediate gets sign-extended. You *can* use the 32b form, of course.

One of the connections my asterisk PBX has is an analog extension from a Comdial hybrid. On the Comdial system, message waiting is turned on by dialing *3 and then the station number. It is turned off by dialing #3 and the station number. I was wanting to have Asterisk (or Comedian mail) set the message lamp in the Comdial system when a new message arrives for a user, and extinguish the lamp

Hi, My Asterisk server process irregularly segfaults, ie. it usually works fine (is stable) when there's low traffic but repeatedly crashes during morning hours when there are more calls. I gdb'ed the core dump files and found that the culprit may be format_mp3. So I disabled MOH today and will see if that's the cause. I know that mp3 files are known to cause * crashes but what I