similar to: Asterisk DTMF 'talkoff' issues

Hi List, I am trying to get DTMF into conference room. for conference I am using Konference module. Konference don't have an option of DTMF gets. Is there any way by which I can get DTMF within conference room? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, I have write down my code on which chanspy not working when I make a group with name of spy. Please help me where is the issue on that. a) caller will call this number to join konference and spy group exten => 43681111,1,Answer() exten => 43681111,n,NoOp(****${CHANNEL}****) exten => 43681111,n,Set(GROUP(${CHANNEL})=spy) exten => 43681111,n,Set(a=${GROUP_LIST(spy)}) exten

HI, I have installed asterisk 1.6.2.18 with konference 1.7, All things are working fine but when we start taking DTMF then key 3 not get my asterisk. When we use landline number(dedicated number) than all DTMF is capture and asterisk work fine. In case of mobile only key 3 don't work. Strange when I use my touch screen number then most of the DTMF digits don't get's my asterisk....

Hi, I have released AppKonference 2.5 today. This release fixes a bug that can cause audio problems when conference frame caching is enabled. It also fixes the spy feature so that more than one spyer can spy on a channel at the same time. If more than one spyer is unmuted, their audio is mixed and whispered to the spyee. -- Paul Albrecht

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( But when I used Vanilla Asterisk then All things are working.... Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf*

Hi List, I want to set my caller ID and name with asterisk. So that when I make outgoing calls then destination end will see my name with number. from asterisk end I set both the things into dialplan. --------------- -------------- exten => _X.,n,Set(CALLERID(num)=9172341457) exten => _X.,n,Set(CALLERID(name)="Virendra Bhati") But when call reach to destination number then only

In an attempt to fix problems with EAGI delays in 1.4 (see my other message for more on that), I've tried upgrading to 1.6, in case it's a bug that's fixed in the newer version. Unfortunately, I'm having all kinds of trouble with this new install. My system relies on conferences, and whenever I add any channel to it (adding a SIP connection, playing an audio file, activating

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( Below are the details of configuration files. Even default MOH is also not working.... *Asterisk Version 1.6.2.17.2 * *1) Extension.conf* [incoming] exten => 6000,1,Answer exten =>

Humm... Seems like my message didn't make it. Here we go again.. /Johan -------- Original Message -------- Subject: Re: [asterisk-users] ControlPlayback's options Date: Sun, 05 Jun 2011 22:19:18 +0200 From: Johan Wilfer <lists at jttech.se> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> On 2011-06-05 19:54, virendra

Hi, I have a requirement where the DTMF entered by a member in konference is passed on to the other members. But the DTMF is not being recognized, when checked the events from manager API, I do see DTMF event being passed, but some how it is not passed to other members. This tells me - may be it is not an asterisk issue, but more a konference application issue. Is this not supported by

Hello I've never used Asterisk for a three-person call, and would like to check that MeetMe is the way to do this. The ADSL modem provided by my ISP offers free calls to landlines/cellphones when using a handset connected to an RJ11 port on the modem. A three-person call can be set up by using the standard PBX sequence: 1. Using the handset, call party #1 2. Hit "R" key on

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

hi,list i installed App_Konference in my Asterisk 1.6.2.11. and i write in dialplan like this: exten => 95040,n,konference(1234,RVxTH) it works fine. but I want to record the conference, if use MeetMe , i can use 'r' option to do this. but there is no 'r' option in konference , Could you tell me how to do this? -- Thanks & Regards Sucan

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Hi List, Is there any way by which we can remove asterisk from machine without deleting folder manually? I did google and gets various solution by no success. even after deleted asterisk will be there ..... ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi , I know it's might not the right way to asking such stupid question. But I want to take help from experts into VoIP fields so I have to decided to post here. Please help me which will be the best VoIP conferencing phone which will cover 10 Persians into conferencing with best audio support ? -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next

Hi Asterisk guys, Flite is not working with asterisk 1.6.2.17. Flite is working with asterisk 1.4. Please help me how to use it with asterisk 1.6 ....... Thanks in advance. ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]:

Hi list, I want ot do basic work (add-edit-delete) into asterisk configuration files, like sip.conf, manager.conf,musiconhold.conf etc. Please guide me how to configure all these files from from AMI connection. I am able to login into AMI from Login action but I want to do more task in to it. *AMI login:- * *login.php* <?php $socket = fsockopen("127.0.0.1","5038",

Hi Team, I have installed asterisk-12 Beta but when I try to asterisk start then get below issue. *[root at cs-gb-pwr-1-04 asterisk-12.0.0-beta1]# asterisk -r asterisk: error while loading shared libraries: libjansson.so.4: cannot open shared object file: No such file or directory [root at cs-gb-pwr-1-04 asterisk-12.0.0-beta1]#* -- Thanks and regards Virendra Bhati +91-9718500594