[Asterisk-Users] point to point T hookup?

[Sean Cook]

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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 router then do QoS
on both routers and do everything VoIP on the asterisk side... Then
you have no hardware necessary for your trunks (other than the routers
of course)


Jonathan Miller wrote:
> 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.
>>>>>>
>>>>>> -A. _______________________________________________
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