Displaying 9 results from an estimated 9 matches for "cbeyond".
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beyond
2003 Dec 02
4
Configuring new system for a non-profit organization
...ims' Assistance fried as a
result of the building power being cycled. I'm now in the process of
building an * system to replace the failed PBX. Minimum cost is the
priority.
I have a T100P card installed in the new system, and I am about to order
integrated T1 services from the "CBeyond" company. They will require
eight voice channels and at least eight data (they presently have DSL.)
The rep. says that for the same cost ($520/mo) they will get all 24
channels as data--the channels are dynamically allocated. As each voice
call is initiated, a channel will be pulled fr...
2006 May 15
2
Asterisk with SIPconnect
Has anyone had any experience connecting Asterisk to Cbeyond's
SIPconnect service (http://www.sipconnect.info)? Any opinions?
Thanks,
-Brian
2003 Dec 02
0
Configuring new system for a non-profitorganization
...> result of the building power being cycled. I'm now in the process of
> building an * system to replace the failed PBX. Minimum cost is the
> priority.
>
> I have a T100P card installed in the new system, and I am about to
order
> integrated T1 services from the "CBeyond" company. They will require
> eight voice channels and at least eight data (they presently have
DSL.)
> The rep. says that for the same cost ($520/mo) they will get all 24
> channels as data--the channels are dynamically allocated. As each
voice
> call is initiated, a chann...
2006 Apr 04
0
Jitter in SIP calls?
I'm experiencing a very strange problem with SIP calls with a CLEC
(CBeyond). The downstream audio with the telephone on mute is
excellent. However, when there is upstream audio (even breathing) from
the mic, the downstream audio is clipped and sometimes dropped.
The strange thing is, if I Monitor the call, the downstream audio in the
wav file is perfect, even though...
2006 Apr 04
0
Jitter in SIP connection
I'm experiencing a very strange problem with SIP calls with a CLEC
(CBeyond). The downstream audio with the telephone on mute is
excellent. However, when there is upstream audio (i.e., breathing) from
the mic, the downstream audio is clipped and sometimes dropped.
The strange thing is, if I Monitor the call, the downstream audio in the
wav file is perfect, even thoug...
2003 Jun 08
10
VoIP Provider
Hi,
I am just about to move out from my parents home and think about how I
will phone from now on. In Germany there is a provider (QSC) who
offers DSL (1024 down/256 up) with fastpath without volume or time
limits.
Does anybody know a comercial (or even semi-professional) provider who
lets me dial out through H323 (or another protocol) and also offers an
number where I can be called from
2004 Dec 17
2
T-1 vs channelised T-1?
OK. Now I show my ignorance.
What's the difference between a T-1 and a channelised T-1? I see that
Covad's voip service (formerly GoBeam) requires a channelised T-1. Then
I read recently on the list that many T-1s being installed are actually
HDSL....which would be not a T-1 at all...right?
Michael
--
Michael Graves mgraves@pixelpower.com
Sr. Product Specialist
2009 Mar 09
0
asterisk-users Digest, Vol 56, Issue 23
This is what you show CBeyond. You have some vouchers there from CBeyond
that will allow me to get paid from them not you.
Chuck Coleman
President CCI Technologies/CC Call Center/CSI Technologies
Director of Managed Services for Gurus2go
Cell 510-439-6501
Confidential Email: This email and any files transmitted with it...
2005 Aug 10
3
SRV implementation supporting priority
Does anyone know when Asterisk is supposed to support prioritization of
the SRV records returned. I think it's accepted that right now the
Asterisk always just uses the first record returned regardless of
priority.
I also noticed, on register, that it queries for an A record first. If
it gets a response, it will use that address to send the REGISTER. If it
does not, then it will launch an SRV