Displaying 7 results from an estimated 7 matches for "codetyr".
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codetur
2007 May 18
0
Re: asterisk-users Digest, Vol 34, Issue 82
...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-Typ...
2007 May 18
0
cpu usage for G.729 codec
...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-Typ...
2005 Mar 08
4
force SIP authentication
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
2005 May 31
3
Opinions of Sphinx?
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
2005 Mar 03
4
[OT] - Why should I answer a Newbie question, therethick!
...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...
2005 Jun 09
23
Voicemail and MS Exchange Synchronization
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
2005 Feb 16
1
Help Please!!!!
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