similar to: Best VoIP conferencing phone ?

Displaying 20 results from an estimated 900 matches similar to: "Best VoIP conferencing phone ?"

2012 Jan 12
1
how to set callerid in php AGI file.
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();
2011 Dec 27
1
how to used SIPp for sip load testing
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
2011 Dec 27
3
how to stop hacking of my server
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
2011 Jun 10
2
How to remove asterisk ?
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:
2011 Apr 22
7
Flite issue
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:
2012 Jul 26
1
Asterisk Realtime issue after registering with x-lite
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]:
2011 Sep 02
5
how to add-edit-delete entery into asterisk conf files
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",
2011 Nov 21
1
video calls not working
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 ;
2013 Oct 21
3
Asterisk-12 issue after successful installation
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
2011 May 17
3
how to know how many calls are on hold
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>
2011 Jun 10
1
Asterisk issue or VoIP provider issue ?
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
2011 Jun 08
2
No IVR listen at device end......SIP phone is working fine
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
2011 Apr 11
1
Asterisk MOH not working with Elastix asterisk 1.6.2.18
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*
2011 Apr 08
2
MOH not working
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 =>
2011 May 26
5
make calls from DID
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>
2011 Jun 07
1
How to get DTMF in Konference module in Asterisk
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:
2011 May 29
3
Why PRI not BRI ?
Hi List, I have stupid question but I want to know it. Why we use the PRI insted of BRI ? Just for the sake of number of lines or any thing else ? And why SIP is used for making calls rather then IAX? Even we know IAX takes 1 channel for making calls? ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Reader -------------- next part -------------- An HTML attachment was
2007 Nov 07
1
Polycom SoundStation VTX 1000 with Asterisk?
Anyone successfully using the Polycom SoundStation VTX 1000 with Asterisk? I can't see any mention of it on the wiki page: http://www.voip-info.org/wiki-Polycom+Phones Thanks, Alvin
2011 Apr 19
1
How to know how many calls are into hold by asterisk command
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
2011 Apr 13
1
How to know extensions status ???
Hi, How to know the all SIP extensions status with AMI's ExtensionState ? What is the value should I pass in Context: <> ?? which will be define at context here ? shell I use sip.conf's context for that extension or any other? extension : <> ?? extension will be SIP/100 or just 100 ?? Please guide me ........... ----- Thanks and regards Virendra Bhati +91-9172341457