similar to: Queue

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 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 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 creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run " show applications" at cli> . my application not displayed. what's wrong??? any suggestion!!! thanks 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 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

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, 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

Hello, everybody Two questions: 1) I am new to maillists, and particularly to r-help maillist. Where specifically do I go online to see my question and answers to it??? If I use the searchable archives, or archives by Robert King, I see my question but not the answers, though I know that at least one persion posted the answer to r-help. Why do not I see the answers? 2) I need to produce 104

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

hi, all I am using asterisk 1.2.12.1 and zaptel 1.2.7 and libpri 1.2.1 version. I have created Ivrs(very big) .It works fine in sip phone , but when i call through zaptel digit sens problem occured. Asterisk doesn't sens any digit pressed.Our pstn is CORAL pbx. any suggesion.. thnks, Bhrugu Mehta(india, gujarat) -------------- next part -------------- An HTML attachment was scrubbed... URL:

HI, all I have tested chanspy app. to monitoring agent and customore conversation. if customer and agent are already in conversation , using spy we can'nt here anything on that extension(agent extension). if next time calls come to that chan we listen that conversation. any idea????? thnks, in advance Bhrugu Mehta

Hello, I was demonstrating Asterisk capabilities with a SIP Soft phone to a Key system installer yesterday, and we were discussing where Asterisk can fit into that market. He brought up some interesting, user-centric questions which I couldn't answer. I didn't find anything in google that realy addressed some of these questions, so I figured I would post them here to see what yall

Maybe don''t make anchor tags draggable? Or you could do something like change the href to point to ''#'' and move the actual href into your javascript, and only call it if they click without dragging. Still, I think making a clickable item draggable is begging for problems. > -----Original Message----- > From:

Hey all, I'm trying to run the D&D Adventure Tools and the D&D Character Generator. Both use the same base program, so I'm getting the same thing whenever I try to run either of them which is the following: Code: Unhandled Exception: System.IndexOutOfRangeException: Array index is out of range. at UpdaterStub.Program.Main (System.String[] args) [0x00000] in <filename

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.

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.