At 08:58 AM 5/10/2004 +0100, Gavin Hamill wrote:
http://www.voiptalk.org/ - this is the service-side of TelAppliant,
official
UK Digium resellers.
I've written to VoIPTalk a couple of times and never got any response from
them, and their outbound calling rates aren't fantastic. I would be
At 12:58 PM 4/5/2004 -0500, Steven Sokol wrote:
I regret that I've only used MeetMe a few times, and only up to two users.
Perhaps others that are using MeetMe could comment on the number of
concurrent conferences and total users they have asterisk running with.
The
specs of the systems
At 12:41 AM 3/20/2004 -0600, Carlos Chavez wrote:
I have been trying out Asterisk with the speex codec with X-lite as a
client. I applied the REG patch on my windows machine that is recommended in
Voip-info.org. Every time I make a call I get the following error:
codec_speex.c:167
I am using an MVP-210 as FXS -- I haven't tried FXO.
Here's my sip.conf entry:
[mvp-x303]
type=friend
host=192.168.1.93
username=303
dtmfmode=rfc2833
context=fs1
disallow=all
allow=ulaw
(Not sure if dtmfmode is correct.)
Username must be an extension number that appears in the MVP210's inbound
I have Asterisk 0.7.2 running on a RedHat 8.0 box. Before installing
Asterisk, I installed libogg-1.1 and speex-1.0.3. speexenc and speexdec
work OK from the command line. I see in Asterisk's startup messages that
it's registered the translators lintospeex and speextolin.
I'm using a
At 07:35 PM 3/9/2004 +, Fran Boon wrote:
On Tue, 2004-03-09 at 19:11, John Chester wrote:
A call from a hardware phone using ulaw to an Xten phone using speex
fails. When the Xten phone answers the call, Asterisk produces an endless
stream of error messages:
WARNING[311313]: codec_speex.c
