[Asterisk-Users] point to point T hookup?

[Jerry Jones]

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 channels as a group and dial by the group number.

Using pri will pass callerid info for you across the connection



On Jun 28, 2006, at 9:30 AM, Jonathan Miller wrote:

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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 kind of T1?  TDM?  Data?  What type of signaling are you  
>> planning
>> to use e&m?  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 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 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
>>>
>>>>> between the two sites, there should be no difference in voice
>>>
>>> quality.  For
>>>
>>>>> a true point-to-point system this is trivial to achieve, and
>>>
>>> maximizes the
>>>
>>>>> bang-for-buck ratio of your interoffice connection.
>>>>>
>>>>> Obviously having two ADSL connections is not true "point to
>>>
>>> point" -- you
>>>
>>>>> will want a leased line, or a dedicated connection to a common
>>>
>>> provider who
>>>
>>>>> has the prioritization of voice traffic in your SLA.
>>>>>
>>>>> You could, in theory, have higher than telco quality voice
>>>>> calls
>>>
>>> with a
>>>
>>>>> VOIP system, as you are no longer restricted to 8kHz-sampled,
>>>
>>> 16-bit audio.
>>>
>>>>> Naturally the phones must support this for this to work.
>>>>>
>>>>>> What configuration areas are there to be set and how are they
>>>
>>> diffent
>>>
>>>>>> than just a standard PRI, which I have working now?
>>>>>
>>>>> If you put a point-to-point DS1 between sites, it's easy.
>>>
>>> Asterisk can act
>>>
>>>>> as a PRI CPE or CO endpoint.
>>>>>
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