[Asterisk-Users] Channelbank Recomendation and GS102 question

Jonathan Moore moorejon at usd465.com
Thu Dec 4 17:16:08 MST 2003


I looked at the same idea, but since we are not in a major metro area we 
couldn't get a mixed data/voice circuit. If a local number isn't necessary (say 
for 800# inbound) you could also go with a VoIP provider like Vonage for PSTN 
connectivity which would also eleminate the need for a channel bank (or even 
the T1 card)
-- 
Jonathan Moore
Director of Technology
Winfield Public Schools
Office 620.221.5100
Fax 620.221.0508


Quoting Walker Haddock <whaddock at datacrest.com>:

> On Thu, Dec 04, 2003 at 06:26:49PM -0600, Paul Oster wrote:
> > Hi All.
> > 
> > I'm working on an * configuration.  We require 8 inbound POTS lines, and 
> > CT1 or PRI seems like it will be
> 
> We have an installation with 9 inbound voice channels (one is the fax) and
> 768K data.  It is a Hybrid PRI.  It terminates into a T100P.  It is working
> great!  The cost was better than the POTS plus data.
> 
> > quite expensive at that level.  I've read that a T1 Channelbank plus 
> > the  T100P would be a (the?) way to go
> > for this situation.  What is the recommended channelbank for use in this 
> > scenario?  From searching the archives
> > I see a lot of suggestions to get "a channelbank" from ebay.  I would 
> > prefer to be able to use new products
> > so I can easily duplicate the setup for other branch offices in my
> company.
>  
> We're not using a channel bank.  I have one port on a TDM card for the fax
> machine.
> 
> > 
> > My second question relatees to the Grandstream phones.  When they are a 
> > member of a queue group, I get a loud
> > annoying ring in the handset when its in use and another call comes in 
> > on the queue.
> > 
> > Is there a way to enforce 1 call per phone in sip.conf?  Either that or 
> > a way to tell the GS102 to return busy when
> > * trys to send them a call.
> 
> You need to use the latest CVS, it includes the work done by Paul Lieu.  It
> works great.  You just configure your sip.conf according to the notes in the
> bug report:
> 
> http://bugs.digium.com/bug_view_page.php?bug_id=0000408
> 
> It solved my GS BT102 call waiting ring in ear problem!
> 
> Walker
> > 
> > Thanks in advance.
> > 
> > Paul M. Oster
> > 
> > 
> > 
> > 
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> 
> -- 
> ********   DataCrest, Inc. -- Technically Superior   ******************
> Walker Haddock                       http://www.datacrest.com
> DataCrest, Inc.                    e-mail:  wh at datacrest.com
> 1634A Montgomery Hwy.    phone:  1-888-941-3282, 1-205-335-8589
> Birmingham, AL 35216                  fax:  1-205-823-7838
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