leg?
Thanks for your help
Robb
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Isaac
be able to dial through asterisk as if it was a PSTN
connection?
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Isaac McDonald
Got VoIP?
[EMAIL PROTECTED]
Cell: +1 253-223-8673
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Anyone know where I can get the patch described here:
http://lists.digium.com/pipermail/asterisk-users/2004-May/045134.html
I am getting compile errors when trying to compile speechtools...
Any help would be greatly appreciated,
Isaac
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Has anyone had any success using the Adit 600 with the CMG card talking
MGCP to asterisk? I want to have a central asterisk server with 10 Adit
600's at various locations providing 24 FXS ports
Thanks,
Isaac
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Does anyone know of a hardware transcoder? Or a software transcoder for
that matter. I would consider using asterisk but it seems that Asterisk
per the WIKI can only support at most 100 channels transcoding from
g.711 to g.729. I would be transcoding from g.711 to g.723.1 or g.729.
Thanks,
As far as a low credit warning is concerned.look at bug id:
1353 on http://bugs.digium.com I have been using it just fine.
usedcanon wrote:
I have a requirement for a setup with prepaid call credits.
I am aware of the two applications available (been researching for the past
week),
Ian McLaughlin wrote:
Has anyone got Asterisk talking successfully to an Altigen PBX using h323?
I can successfully make calls from Asterisk to Altigen, but calls from
Altigen to Asterisk get a fast busy.
There seems to be a lack of h323 example files (or maybe I'm looking in the
wrong places) as
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need this functionality before I can migrate to * completely.
2) Are there currently
It works now! I did nothing on my end either. VP must monitor this list.
Isaac
Robert Jackson wrote:
Just a quick couple of questions for ya'll.
1) Does anyone know if VoicePulse Connect will be supporting dtmf tones?
I have had a terrible time getting a hold of anyone over there, and I
need
I have set up my * box to provide free calling. You can access it by
dialing 1-700-945-2475. Once you hear the prompt dial 1 the area code
and number. I would also like to test some direct incoming IAX
connections from some other * boxes to see if I can terminate PSTN calls
that way. If you would
That did the trick! Thank you Christopher!
Isaac
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Christopher
Stephens
Sent: Wednesday, October 29, 2003 11:39 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Voicepulse and IAX
The instructions they
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