similar to: application not load

Hi, all What is the use of astdb? Is it used to store realtime values like sip etc. Regards, Bhrugu Mehta -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100127/b13d6a62/attachment.htm

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

hi, all i want to create cd-rom with asterisk. how it possible. when i put disk in cdrom it boot automatifcally and auto-start installation like TRIXBOX. any idea. thnks, Bhrugu Mehta

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

hi, all I am new to zaptel programming. can anybody help me how to start this. or any ref. site or matirial availabel. i want to use c lang. for this. thnks in advance. Bhrugu Mehta

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in

hi, all is there any way to send image on sip channel ? Regards, -- Bhrugu Mehta Sr. S/W Engineer (D&D) VOIP,Telephony Team India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100317/3cb8322a/attachment.htm

hi, all I want to connect asterisk with oracle database. how to start this , that's i dont know . any pls help me thnks in advance Bhrugu mehta

hi, all Is ther any way to set 3-way conference using queue app other other way using queue app. scenario: custmore call to queue , agent answered than agent transfer to third persion, so the three call communicate with each other. Regards, -- Bhrugu Mehta Sr. S/W Engineer (D&D) VOIP,Telephony Team India -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, The asterisk book states the following for using ODBC to connect to an MS database. ? The pooling and limit options are quite useful for MS SQL Server and Sybase databases. These permit you to establish multiple connections (up to limit connections) to a database while ensuring that each connection has only one statement executing at once (this is due to a limitation in the protocol

playing with this for two days, so don't jump too hard, gurus.) _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of bhrugu mehta Sent: Monday, January 25, 2010 6:11 AM To: asterisk-users at lists.digium.com Subject: [asterisk-users] queue Hi, all Is ther any way to pass channel queue such a way

hi, all I am new to use DeadAgi, can anybody help me how to use DeadAgi, actually i want this, when caller hangup his/her phone, i want to send packet to my other app that check caller hung up done.

I successfully obtained the Asterisk code and extracted them into /usr/src. When I make and install asterisk, zaptel, libpri etc. Are they supposed to move automatically into their respective directories? I cannot find: /etc/asterisk/ /usr/lib/asterisk/modules/ /var/lib/asterisk Do I have to manually create them or this is failed install? Thanks. -------------- next part -------------- An

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

Hello, Are there any other other voice changer applications to Asterisk other than the one from Lobstertech? (http://lobstertech.com/voice_changer.html) Specifically interested in open-source but can have a look at economical commercial alternatives as well. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I'm using Grandstream IP phone GXP2000, with Asterisk 1.4.15 I'm using g729 codec and want to use only this codec for the calls. My normal calls are going fine. But issue is coming when I'm using the conference from the Line1 and Line2 Option. When I'm initiating the conference at that time, IP phone is sending the G711ulaw for the conference call, while in my phone I've

Hi, I' still fighting the problem, that I can talk from one SIP phone to another, but I can't hear the output of the playback or similar applications: exten => 202,1,ANSWER() exten => 202,2,PLAYBACK(tt-monkeys) exten => 202,3,HANGUP() When I dial 202, asterisk show the following on the cli: -- Executing [202 at local:1]

Hi List, I wanted to know if anyone knew of a way with asterisk to "switch the voice" of a caller from male to female or vice versa. Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070426/2d483875/attachment-0001.htm

hi, all I have problem to sense digit in my ivrs. scenario is below: I am using zaptel T410P digium card to competible with my PSTN(CORAL) [ivrs] exten => s,1,Background(welcome-ivrs) exten => 1,1,Playback(welcome) exten => 2,1,Playback(goodby) sound file are .wav files. when i dial no. from analog phone to launch ivrs welcome-ivrs.wav file plays and when i press digit 1 play wecome

hi, all actually i can't understand what is the use of autoservice.c file. can anybody help me. thnks in advance. Bhrugu mehta