similar to: AGI Pascal

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: Is DynExtebDB module still working?? -- JO

Hi Guys. I've had a lay off from Asterisk for 12 months but I am starting to look into it again. I am not very Linux savvy and found it hard going the last time. I've started playing with it in the last 3 weeks and I have to admit to making more head way this time. The first problem I'm stuck on and I cant find a solution to is that sound files that I have recorded (be it by

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

I'm a total newbie at this telephony stuff but I'm putting together a low cost PBX for my small company and wanted a check on the h/w I'm planning on ordering and my system configuration. Any input is appreciated. Take it offline and email me directly if appropriate (mstupak@comcast.net). Here's what I'm planning: === Parts List === 1 Digium Wildcard TDM400P w/

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, 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 --------------

Does anyone know if the E1 cards that digium sells work in India. Also are there any distributers for those cards in India. By E1 cards I mean E100P, TE410P or TE405P -- regards Vikram (http://www.vicramresearch.com)

How do you prepend. I want to be able to dial 7 digits instead of of 11 for local calls. Can someone post there extensions.conf part that is relavent?

I am trying to set up the Message Waiting Indicator (stutter tone/light) so that my cisco ata-186 will let my phones know there is a message waiting. However this does not seem to be very well documented. I found this on wiki mailboxnumber@context ... where does that go? Do I put it in my SIP.conf definition for my cisco ata, or where. In my SIP cisco definition i already have a

Hi I've need to reinstall my asterisk software (hard drive failure). I'm back and running to a make samples state. I have backed up all of my conf files (ok so they were about a week old but much better than starting from scratch), the problem I am having is with WS_FTP Pro. Basically I used to connect to my asterisk server using this software no problems just using root as

Hi, As you asked, I have included my diff to what I did for the DIAL command. I probably didn't stick to some * pre-agreed standard of coding or something, so if these things offend you then I suggest that you close your eyes. :) The biggest thing to consider when you are doing a prepaid system is, what if the person with the same account in/out calls twice? I chose, for now, just to keep

i tried asterisk ilbc codec against kphone. when the call got connected, i started to immediately get these kind of message to the console: WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of 52 bytes long from RTP (50)? WARNING[14350]: File codec_ilbc.c, Line 141 (ilbctolin_framein): Huh? An ilbc frame that isn't a multiple of

-------- Original Message -------- Subject: Re: [Asterisk-Users] Troubles with Kphone Date: Tue, 25 May 2004 15:44:15 +0530 From: Murali Krishnan <murali@bksys.co.in> Reply-To: ismk@myrealbox.com Organization: bk SYSTEMS (P) LTD., To: asterisk-users@lists.digium.com References: <200405250652.46370.klky3@fibertel.com.ar> enano wrote: >Hi , > > > >I'm triying to use

hello, I am running asterisk on one linux PC and want to talk through this server using Kphone installed on 2 different PC's. These are the extra lines added to sip.conf and extensions.conf respectively. sip.conf [jitha] type=friend host=dynamic secret=jitha context=sip dtmfmode=inband [sudhananda] type=friend host=dynamic secret=sudhananda context=sip extensions.conf [sip]

Hello, I know that my Kphone question may be a bit off topic, but I have been busy with this again and again for about one month now, sent three mails to kphone@wirlab.net (the contact address mentioned on http://www.wirlab.net/kphone/index.html), asked for a solution in a german ip phone forum and tryed many things by myself. I try to compile KPhone 4.0.3 (tryed CVS Version as well) but

Is there a package for CentOS of GNU/Pascal? -- 21:50:04 up 2 days, 9:07, 0 users, load average: 0.92, 0.37, 0.18 --------------------------------------------------------- Lic. Mart?n Marqu?s | SELECT 'mmarques' || Centro de Telem?tica | '@' || 'unl.edu.ar'; Universidad Nacional | DBA, Programador, del Litoral |

Is anyone successfully using call queues and call groups? If so do you have an example configuration? The wicki and mailing list information I found is pretty old. Thanks! Paul pmahler@signate.com

Im new to Astererisk. I compiled the latest CVS and setup the server. It looks like things are working. I'm running kphone, x-lite and sjphone to test things out. The kphone (local to the asterisk server) can call and receive calls from any of the 2 windows machines. The first windows phone I start I can send/receve calls the second one I cannot. I. No matter which one I start first only

I just can't get kphone to register with asterisk, i can make calls to the demos and even get into the mailbox but kphone cannot register. Here's my story. Can you help me?? Please I have installed asterisk on debian using apt-get install asterisk. I have configured an extension in extensions.conf as follows exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt) exten =>