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...SIP subscribe for peer without mailbox: 1000 Really destroying SIP dialog '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method: SUBSCRIBE If anyone have any suggestion please reply to me. -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120726/b5e75cca/attachment.htm>

...wr-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 +91-9250078532 E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/2013102...

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 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 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, 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 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 =>

...an I > > implement for outbound fax in this case. > > > > Please advice. > > > > -- > > Regards, > > > > Ahmed Munir Chohan > > > > > Thanks and regards > > Virendra Bhati > +91-9718500594 > Asterisk Developer > E-mail-: virbhati at gmail.com > Skype id:- virbhati2 > New Delhi(India) > [image: View my profile on > LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755> > -- Regards, Ahmed Munir Chohan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://li...

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",

...; <div> <div> <div> <p class=3DMsoNormal style=3D'margin-bottom:12.0pt'><font size=3D3 face=3D"Times New Roman"><span style=3D'font-size:12.0pt'><br> On Apr 11, 2011, at 6:24 AM, virendra bhati &lt;<a href=3D"mailto:virbhati at gmail.com" = target=3D"_blank">virbhati at gmail.com</a>&gt; wrote:<o:p></o:p></span></font></p> </div> <blockquote style=3D'margin-top:5.0pt;margin-bottom:5.0pt' type=3Dcite> <div> <div> <p class=3...

I've been using Gmail and thought you might like to try it out. Here's an invitation to create an account. You're Invited to Gmail! virendra bhati has invited you to open a Gmail account. Gmail is Google's free email service, built on the idea that email can be intuitive, efficient, and fun. Gmail has: *Less spam* Keep unwanted messages out of your inbox with Google's

Hi List, I am confuse about this feature. When we use Wait(20) in active call session then it's work. But when we use after hangup the call then Asterisk don't wait from define time. Ex:- [call_log] exten => 4368,1,Answer() exten => 4368,n,Flite("Welcome") exten => 4368,n,Set(__StartTime=${STRFTIME(${EPOCH},Asia/Calcutta,%Y-%m-%d %H:%M:%S)}) exten =>

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:

...nt to know which events is regenerates(activate) when asterisk manager logoff from asterisk. I saw *fullybooted *was active when I login into asterisk as manager but no event at logoff time i found. -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 Software Developer E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/2013101...

...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 +91-9250078532 E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131021/3b8e6ebc/attachment.html>

...t; [Context] => from-sip [Exten] => 1234 [Priority] => 3 [Uniqueid] => 1382599433.22 ) Please check the BOLD section. earlier is was *[CallerIDnum] * *So 'n' is now 'N' * -- Thanks and regards Virendra Bhati +91-9718500594 +91-9250078532 Asterisk Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) [image: View my profile on LinkedIn]<http://in.linkedin.com/pub/virendra-bhati/6/a30/755> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/2013102...

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

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 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

Hi List, I am using Asterisk 1.6.2.18. One strange problem come into my knowledge after using this version of asterisk. Without pressing any digits or key from my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and