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...c('Dial',"SIP/00918885268942 at voipon,60,r"); ?> And CLI> == Using SIP RTP CoS mark 5 -- Executing [101 at outbound:1] Answer("SIP/2209-000026d3", "") in new stack -- Executing [101 at outbound:2] AGI("SIP/2209-000026d3", "/home/virendra.bhati/outdial.php") in new stack -- Launched AGI Script /home/virendra.bhati/outdial.php <SIP/2209-000026d3>AGI Tx >> agi_request: /home/virendra.bhati/outdial.php <SIP/2209-000026d3>AGI Tx >> agi_channel: SIP/2209-000026d3 <SIP/2209-000026d3>AGI Tx >> a...

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 innovative technology. *Lots of space* Enough storage so...

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110610/610edf28/attachment.htm>

...t 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 part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111130/c8a6e992/attachment.htm>

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110422/d183d59c/attachment.htm>

...811]: chan_sip.c:16897 handle_request_subscribe: Received 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>

...;); fputs($socket, "directory: /var/lib/asterisk/moh\r\n"); fputs($socket, "Reload: yes\r\n"); fputs($socket, "ActionID: 9873497149817\r\n"); fputs($socket, "Action: Logoff\r\n\r\n"); ?> After doing all no success :(( ----- Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110902/965fcc13/attachment.htm>

...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 +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....

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>

...ontext 'default'. == Using SIP RTP CoS mark 5 [Dec 27 17:37:55] NOTICE[28001]: chan_sip.c:20785 handle_request_invite: Call from '' to extension 'service' rejected because extension not found in context 'default'. haddock8-astrx*CLI> -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111227/b4d08203/attachment.htm>

...y 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 number is display, name was display as *unknown * Is this issue of voip provider or Asterisk 1.6.2.18 ? I contact them they replay me that it's your end issue not my end..... ----- Thanks and regards Virendra Bhati +91-917...

...ne 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 +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110609/efd31252/attachment.htm>

....246>' failed for '62.141.54.169' - Wrong password [Dec 26 21:21:21] NOTICE[1770]: chan_sip.c:22318 handle_request_register: Registration from '"4411" <sip:4411 at 204.152.194.246>' failed for '62.141.54.169' - Wrong password -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20111227/f6c9afa7/attachment.htm>

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: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110607/2c8140e9/attachment.htm>

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 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 scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110529/f591743f/attachment.htm>

...0000006", "CHANNEL(musicclass)=BSNL") in new stack -- Started music on hold, class 'BSNL', on SIP/1001-00000006 -- Stopped music on hold on SIP/1001-00000006 == Spawn extension (bhati, 6000, 4) exited non-zero on 'SIP/1001-00000006' ----- Thanks and regards Virendra Bhati +91-9172341457 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110408/6f72dfca/attachment.htm>

...; > good i.e. later forward the fax via email but don't know how can 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...

...onState ? 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110413/8c1c3906/attachment.htm>

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 was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110419/a8092da3/attachment.htm>