search for: bhrugu

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

...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 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

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 Queue(SIP/1001&SIP/1002&SIP/1003) thanks, Bhrugu Mehta ------=_NextPart_000_002A_01CA9D99.AAFDFE50 Content-Ty...

...st 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

...ly 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 advance. Bhrugu mehta

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, 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

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20100716/d9bed4ea/attachment.htm

....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 and 2 play goodby. but some time asterisk server doesn't sense which digit i pressed so welcome-ivrs file continue with playing. Doesn't stop playing thnks, regard Bhrugu Mehta(SIS)

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 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: http://lists.digium.com/pipermail/asterisk-users/attachments/20080112/d268137b/attachment.htm

...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

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.

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

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]