search for: codetyrant

...xtension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Thu, 17 May 2007 11:22:17 -0400 > From: "Race Vanderdecken" <asteriskusers@codetyrant.com> > Subject: RE: [asterisk-users] cpu usage for G.729 codec > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <01d101c79897$2c802df0$0e01a8c0@PressonMobile1> > Content-Type:...

...xtension AGI to play a file pre-encoded in G.729, do I need a codec? Where is the SW that encodes files in G.729? On Thu, 2007-05-17 at 08:38 -0700, asterisk-users-request@lists.digium.com wrote: > Date: Thu, 17 May 2007 11:22:17 -0400 > From: "Race Vanderdecken" <asteriskusers@codetyrant.com> > Subject: RE: [asterisk-users] cpu usage for G.729 codec > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <asterisk-users@lists.digium.com> > Message-ID: <01d101c79897$2c802df0$0e01a8c0@PressonMobile1> > Content-Type:...

Hello, is it possible with Asterisk to force SIP authentication? Right now, it seesm that just any SIP client can at least connect to my PBX, which I don't want. I want users to authenticate with username and password and otherwise deny them access. Thanks Florian

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

...answers. I am only asking for someone to show me the site and maybe a few pointers on how to start it up. Only because I don't have the time or experience to do it quickely enough to get the newbies off the list. And I am a bit slow with apache and web type sutff, as you can tell by my website codetyrant.com. I will personally pay for the hosting of the list. It is not that I am tired or will ever grow tired of passing out fish and giving fishing lessons it is just I don't have the good fortune to be adept at web interfaces. Also, suggestions for the domain name would be welcomed. Race &qu...

We have a customer considering migrating from a large Nortel PBX with a third-party voicemail system to Asterisk but one of the features they really like is the automatic synchronization of voicemail between Exchange and their voicemail system -- delete a message from the voicemail system and it is deleted from their email inbox and vice versa. Searching has not revealed anything like this

I have a asterisk server with 6 Cisco ATA connected in SIP. My problem is that one of them is dropping calls an I can't figure out what is the problem; I had made a SIP DEBUG PEER 1088 that is the peer with the problem. Any help will be appreciate Thanks Erick Weber VoIP*CLI> sip debug peer 1088 SIP Debugging Enabled for IP: 201.133.170.82:5060 Peer RTP is at port 192.168.1.69:0 Peer