similar to: Why PRI not BRI ?

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 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 list, please help me how to know how many calls are on hold..... -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110517/09cbc325/attachment.htm>

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();

I am going to try and use a passive AVM fritz BRI card for my * setup.. Here is the thing.. I need to order my BRI from BT.. The service that looks to be the one to use is what they call ISDN 2e becasue this has the option to setup hunt groups across multiple ISDN2e lines so I could add another line later to get 4 channels.. According to the BT website in order to use the hunt grouping across

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

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 List, When we make calls into asterisk with the help of our mobile, landline number, Cisco 79XX series then we didn't able to here any IVR which is playing into asterisk server. But when we dial from SIP softphone then all is working fine and we are able to here the IVR sound files. What is the problem in this case please help me.. -- ----- Thanks and regards Virendra Bhati

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 , 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 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]:

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 =>

How to make outgoing calls from DID and what is theway to get incoming calls from DID. -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110526/3f19091d/attachment.htm>

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

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 installed SIPp into my server. But not able to used it properly. how to configure with my server ? how to see logs on webpage ? how to start call testing .... when i start SIPp then found verious hits on myserver. *CLI:- * [Dec 27 17:37:54] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not

Hi list someone is trying to hack my server . Is there any way by whcih I can stop hacking of my server except iptables ? I want to stop on the basis of sip.conf account only. bcoz I can't apply iptables rules on server it's remote server of server provider and we used it for making voip call for customers. for the time been i have close all sip accounts. but can't stop for more then

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

> Thank you all for valuable input, > > another question: when do I actually need the echo cancellation > (hardware / on board /on module ) ? > It depends on your environment. If there are still analog devices in addition to VoIP, I'd say always, but Asterisk has a rudimentary echo canceller already on board. The Telcos use echo cancellers themselves, but it cannot hurt to

Hi All, Is it possible o know how many call are into hold ? who are on hold ? By whom these extension are on hold ? And after 60 sec asterisk will call them automatically as Call Parking do? I wan to make this concept to my PBX system... Thanks in advance -- ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment