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Hi Peter,<br>
<br>
The scenario is like this:<br>
<br>
<a class="moz-txt-link-abbreviated" href="mailto:sip-many-ids@iptel.org">sip-many-ids@iptel.org</a> flux(Internet) --> Asterix -->
TE410P(E1)-->PBX-->regular-office-phones.<br>
<br>
I want PBX to use another(the actual) E1 line for normal PSTN
dialin/dialout calls but also with the other dedicated E1-TE410P line I
want to be used for the same office phones for VOIP dialout/dialin
calls(with assigned SIP ids by Asterix together with PBX). Is this
possbile or am I doing a conception mistake ?<br>
<br>
Regards,<br>
Sergiu<br>
<br>
<br>
<br>
<br>
Peter Svensson wrote:
<blockquote
cite="midPine.LNX.4.44.0411031543390.1475-100000@cheetah.psv.nu"
type="cite">
<pre wrap="">On Wed, 3 Nov 2004, Sergiu Dunca wrote:
</pre>
<blockquote type="cite">
<pre wrap="">We have a E1/digital PBX in the office for ISDN-PRA telecom access and
for digital phones connected to it. So the PBX is E1 capable.
I wander if we can use a E1 interface in the PBX and a Digium PCI card
to interconnect the PBX with Asterix. Then we would like to configure
the PBX and Asterix to assign to the digital normal telephones SIP ids
and to use them by calling over SIP (ex. iptel.org).
</pre>
</blockquote>
<pre wrap=""><!---->
I'm not quite sure which of the following scenarios you are after:
pstn <--E1 Isdn--> Asterisk <--E1 Isdn--> PBX
or
pstn <--E1 Isdn--> PBX <--E1 Isdn--> Asterisk
Actually, both should work fine. For the first case you need something
like a TE405P or TE410P card where you will use two of the four interface.
The setup will to a large extent be in Asterisk. For the second case you
will need a more capable PBX with two E1 interfaces and the configuration
changes will be split between the PBX and Asterisk.
For the first case Asterisk will be more or less transparent to the PBX.
For dial-in only sip access the setup will be easy. Make the incoming
calls on the two pri lines connect to the other pri line and connect the
incoming sip calls to the PBX E1.
If you want to use sip for outbound calls as well you will have to select
a way of deciding which calls are to be sent via sip and which are to be
sent via the isdn connection.
Peter
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</pre>
</blockquote>
<br>
<br>
<pre class="moz-signature" cols="72">
</pre>
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