Thank you for your help Steven.
> Message: 8
> Subject: Re: [Asterisk-Users] Drops due to codecs?
> From: Steven Critchfield <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Date: 03 Jul 2003 15:26:45 -0500
> Reply-To: [EMAIL PROTECTED]
> On Thu, 2003-07-03 at 15:
Hello,
It is my understanding that on the softphone side, asterisk is only
responsible for establishing the session between two phones. If this is the
case, does it matter what type of audio codecs the two phones are using? And
if it does matter, are there any codecs that cause problems wi
Moshe,
I was having the same problem with my software only asterisk pbx setup. I
was using two kphones on different machines, connecting through a machine
running asterisk. They would connect just fine, but voice was not getting
routed through. I installed linphone, which can be found at
I'm trying to get two linux machines with kphone-3.11 two communicate with
each other over asterisk. I have them configured correctly on asterisk to use
sip channels, but when I call from one phone to the other I don't any voice
communication between the phones. According to the phones I'm con
Hello,
I'm new to Asterisk, and am trying to get the basic features under my belt
until I move on to the more advanced ones. Currently I have two softphones
registered with my * server on my network, and one of the phones can call the
other just fine, but when I try to call from the other p
To James, Robert, Woody, and last but not least, Leo. Thank you very much for
your suggestions on Zaurus mic/headphone configurations and the link for the
softphone apps. Your help is much appreciated.
Daniel
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