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Hello Everybody, I was going thru the C code of Asterisk. Does anybody know how does one go about modifying the C code of Asterisk? Please do reply. Regards, Bharat M. Sarvan EZZI BPO Pvt Ltd., PUNE. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050525/3b2a34f4/attachment.htm

...t here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response Regards Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050317/8ccdfc39/attachment.htm

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Hi, I have a little problem running Asterisk. I just got the asterisk, zapttel and libpri sources from cvs. I built and installed it. Next I installed the sample configuration. The problem arise when I try to start Asterisk. Running asterisk -vvvvc I get the following error [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf':

...fig_mysql.c:588 mysql_reconnect: MySQL RealTime: Failed to connect database server asterisk on. Check debug for more info. Does anybody know how to go about configuring Asterisk to access extensions from the mysql database? Please kindly do reply Regards, Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050513/88cab5e2/attachment.htm

...ows AGIException { SIPPeersAction showpeers; showpeers =new SIPPeersAction( ); } } Does anybody have any idea as to where might the problem be..? Please do reply Regards, Bharat M. Sarvan Software Engineer - VoIP EZZI BPO Pvt Ltd., PUNE. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050804/5fb0a1b3/attachment.htm

...any probs, but the control is not getting transferred to the context "incoming-next" upon execution of the Goto statement. Could anybody suggest me as to where might the problem be and any way to get rid of this problem. Please do reply.. Regards, Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050519/50121954/attachment.htm

...RIMARY KEY (`name`) ) TYPE=MyISAM; And there was no dial tone on the phone. Can you please tell me as to what's wrong in the configuration and how as to how does asterisk fetch the details of the peers from the database. Thanks in advance. Regards, Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/fa8db447/attachment.htm

I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call in and through successfully. I was trying to set up the Realtime - picking the sip.conf and extensions.conf from mysql. I was going through some wiki pages, but what i don't understand is - which configuration change makes asterisk stop looking at extensions.conf and sip.conf for sip peers and pick the same from database.

...ve gone through the information given on the site www.voi-info.org <http://www.voi-info.org/> , but didn't get thru. Would you please tell me as to how do I go about configuring the asterisk realtime extensions. Thanks in advance. Regards, Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050517/03abc7f3/attachment.htm

Hi .. When I use the GoTo statement in realtime to goto a priority only ... E.g. Goto(3) then there's no problem But, If I try to jump to another context ... E.g. Goto(othercontext,${EXTEN},3) then it doesn't work If I process the same statement in extensions.conf things go well Are there things broken regarding GoTo in combination with Realtime Extensions ?

...f yes, could you please tell me as how does one go about modifying the code and also which are the files that are modified? It would be very kind of you if could mail me a sample file with the changes made and the original file. Kindly please do reply. Thank you. Regards, Bharat M. Sarvan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050523/44e416ea/attachment.htm

...; at 10.111.16.149 port 5060 expires 120 -- Saved useragent "SJphone/1.60.289a (SJ Labs)" for peer bharat Jan 13 10:08:58 WARNING[1147]: res_config_mysql.c:331 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Thanks in advance Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or...

...But the problem is when I place a call, after executing the last application in the dial plan I get a 403 Forbidden Response from the Asterisk server. Any ideas about the cause for getting the 403 Forbidden Response from the Asterisk Server..? Thanks in advance Thanks and Regards, Bharat Sarvan The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or...

...'s Topics: > > 1. RE: Welltech USA? and Wellgate Products? (kevin ling) > 2. RE: Connecting to live calls (Wai Wu) > 3. RE: Web based SIP client (kevin ling) > 4. Re: 911 and ISDN PRI (Darren Nickerson) > 5. Asterisk returning 403 Forbidden response > (bharat.sarvan@wipro.com) > 6. RE: Connecting to live calls (Alexander Lopez) > 7. TDMoE (Mike Hammett) > 8. SIP-H323 Help and Multiple Listening Port (Kenige Ho) > 9. RE: TDMoE (Alexander Lopez) > 10. Re: Mitel 5220 IP phones (tracinet) > 11. Polycom dialplan restriction (Carlos Chave...

...1. RE: Welltech USA? and Wellgate Products? (kevin ling) > > 2. RE: Connecting to live calls (Wai Wu) > > 3. RE: Web based SIP client (kevin ling) > > 4. Re: 911 and ISDN PRI (Darren Nickerson) > > 5. Asterisk returning 403 Forbidden response > > (bharat.sarvan@wipro.com) > > 6. RE: Connecting to live calls (Alexander Lopez) > > 7. TDMoE (Mike Hammett) > > 8. SIP-H323 Help and Multiple Listening Port (Kenige Ho) > > 9. RE: TDMoE (Alexander Lopez) > > 10. Re: Mitel 5220 IP phones (tracinet) > > 11. Polycom dial...

In all the information on Asterisk it takes about TDMoE to link asterisk servers together. Is this IAX??? How would I use TDMoE. Maybe my first question should be, What is it??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030516/cd74bddb/attachment.htm

....digium.com> Message-ID: <00c301c55c92$e7051de0$0300a8c0@mozart> Content-Type: text/plain; charset="iso-8859-1" HI, The problem is that you are using: incoming-next,60069,1 Use: incoming-next|60069|1 instead RG, Gentian ----- Original Message ----- From: Bharat M. Sarvan To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: asterisk-dev@lists.digium.com Sent: Sunday, April 17, 2005 11:52 AM Subject: [Asterisk-Users] Asterisk real time extensions problem... Hello everybody, I have setup asterisk real...

Skipped content of type multipart/alternative-------------- next part -------------- _______________________________________________ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev