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I have an installation where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead
On Wednesday 28 June 2006 08:48, Jonathan Miller wrote:
An alternative is to put a router and switch at each end and extend a data
network to the other site for SIP traffic. Would that result in better
quality calls?
If you can ensure that voice traffic has top priority in all the routers
Hi Jonathan -
I have an installation where I'll have a site to site data DS1 for use between
two corporate offices. We'll have one asterisk server at each office. I'd
like to be able to route calls over the 24 channels on that DS1 between the
offices, instead of over the voiceT at each location
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I have a true leased line (a T1) between the two sites.
What parts do I configure for Asterisk to utilized the link bi-directional?
On Wednesday 28 June 2006 09:09, Andrew Kohlsmith wrote:
On Wednesday 28 June 2006 08:48, Jonathan Miller
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What kind of T1? TDM? Data? What type of signaling are you planning
to use em? There is a lot of information that that question is
lacking for anyone to advise you ...
Jonathan Miller wrote:
I have a true leased line (a T1) between the two
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Your response leads me to further question this setup...
It's a full data T that is not provisioned.
Being that I control the termination at each end, do I get to specify the
encoding?
On Wednesday 28 June 2006 10:17, Sean Cook wrote:
What
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Typically with a data t1 you are running either HDLC or PPP on either
end. I assume you have a cisco router on either end? Or are you
planning to plug asterisk with a Digium/Sangoma/Other T1 card?
Personally if it is a data t1 I would use a cisco
Assuming it is a dedicated private line p2p T1
Assuming that 23 calls at one time is sufficient
Install a T1 card in each server, plug the T1 in and set one end ofr
pri net, the other for pri cpe.
zaptel.conf and zapata.conf are the files you are looking for. Just
define the 23
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It always helps to read the original post... so I apologize. I think
what you are looking to do is route the calls over the existing data
t1 in which case all you need to do is create an IAX trunk between the
two asterisk servers addressing their