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...which i can do if i uninstall it. How can i make simultaneous calls with the codec installed? Thanks and appreciate your response. Regards, Naveen.Palani Quinnox Consultancy Services Ltd | Pune | INDIA | Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 | Fax : +91 20 4015 2305 | Email : naveenp at quinnox.com<mailto:naveenp at quinnox.com> ________________________________ ?Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO?9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI...

...this bug are they talking about. We had issues with Background syntax working in macro. Is this the bug fix they are talking about. Regards, Naveen.Palani Quinnox Consultancy Services Ltd | Pune | INDIA | Tel : +91 20 40152300 Ext : 316| Mobile : +91 9960466622 | Fax : +91 20 4015 2305 | Email : naveenp at quinnox.com<mailto:naveenp at quinnox.com> ________________________________ ?Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO?9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI...

Hi, I have the asterisk-1.4.11 set up installation on my Ubuntu machine. When i try making a simple incoming call using xlite softphone. I get the following message when i try calling to the number. *CLI> [Feb 19 13:35:40] NOTICE[4137]: chan_sip.c:5331 process_sdp: No compatible codecs, not accepting this offer! Which codec is it talking abt here. How can i resolve this. My dialplan is as

Hi, I have been working with asterisk to make ivr calls (outbound and inbound). I have the functionality - Read(variable|file_name) used in my dialplan. Now i need to pass the variable to my ruby file to compare the data entered with the database (mysql). How can i pass the arguments from my dialplan to the ruby file. Is there a way i can do it with the agi script? Any one has any clues on

Hi, I am using Asterisk-1.4.11 version to make outbound calls and deliver the swift text to audio. My functionality is as for example i make this text to audio deliver the person called. Eg. swift -o /tmp/test.wav -p audio/channels=1,audio/sampling-rate=8000 "Press 1 to confirm. Press 3 to cancel." extension.conf dialplan: [dialout] exten =>

...nd no different model numbers that indicate included optional modules). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - "The Genuine Asterisk Experience" (TM) ------------------------------ Message: 5 Date: Wed, 16 Jan 2008 13:11:26 -0600 From: Naveen Palani <naveenp at quinnox.com> Subject: [asterisk-users] asterisk to mysql database! To: <asterisk-users at lists.digium.com> Message-ID: <00ed01c85873$979cfb30$2505280a at quinnox.corp> Content-Type: text/plain; charset="utf-8" Hello, Is there a possibility to connect from asterisk to...

Hi, I have been using asterisk-1.4.17 version. Have a SIP registration from bandtel sip providers. Use DID numbers for the incoming calls which works fine when i dont use any peer setting in my sip.conf file. But when i use a peer and make calls thru the DID number it doesn't reach asterisk at all. Doesnt give me any errors as well. peer in my sip conf is as given below: [proxy2_bandtel]

Hi, I have been working with Asterisk for the ivr functionalities in the past. I am interested to implement the Jabber - Gtalk in asterisk. For which i installed the iksemel but this didnt help me out. I couldnt find the res_jabber.so file any where in the asterisk source directory. Still when i run the command "make menuselect" the channel driver "chan_gtalk" shows xxx