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stotaro at totarotechn... Guest
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Posted: Fri May 16, 2008 9:45 am Post subject: [asterisk-users] Connecting a PSTN gateway to Asterisk using |
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On Fri, May 16, 2008 at 10:28 AM, Pascal Maugeri
<pascal.maugeri at gmail.com> wrote:
Quote: | Hi
I have a system (S) that has a PSTN gateway to accept incoming calls and
setup outgoing calls from/to Telco network. In the other hand I have a
distant Asterisk box (A) that I would like to connect to (S) using the PRI
interface.
I understand that the proper way is to order to my Telco two PRI lines one
for (S) and another for (A), and configure (S) and (A) to call each other
numbers when they have to interconnect.
Now, might it be possible to connect directly (A) and (S) using their PSTN
interfaces without having to go through to my Telco ?! Does it make sense ?
Is it technically feasible ? I guess that the Telco network is providing
routing, number assignation, etc. and it sounds pointless to do this.
Nevertheless could you confirm it is possible/impossible and why ? Is there
a better way to do that ?
Thanks in advance,
Pascal
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You could price out a point to point T1 between A and S and then send
the traffic over via VoIP. You could use LCR hop on and hop off.
Provide your own routing and number assignment (beyond what is
required by the telco.
I guess it comes down to the cost of the point to point. The calls
between sites would be free as they are VoIP.
You could also eliminate getting the second voice T1 at the remote
location if you don't really need all those channels or you could get
the second T1 in the first site and use Asterisk to handle routing for
local or calls to go over the point to point connection.
Some people will say you can just go over the public internet, but I
don't recommend it. It "may" work OK but what if it doesn't? What
are your calls worth?
Thanks,
Steve Totaro |
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bwentdg at pipeline.com Guest
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Posted: Fri May 16, 2008 9:48 am Post subject: [asterisk-users] Connecting a PSTN gateway to Asterisk using |
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This is 'basically' a tie-line between the boxes.
Yes - it is done all the time between PBX's. You are basically nailing
up a circut between the boxes.
It could be a simple as a simple POTS leased line or a multi-t1 bundle
between them.
How it is physically done with DIGIUM's boards under * ?
Someone else will have to answer that
Pascal Maugeri wrote:
Quote: | Hi
I have a system (S) that has a PSTN gateway to accept incoming calls
and setup outgoing calls from/to Telco network. In the other hand I
have a distant Asterisk box (A) that I would like to connect to (S)
using the PRI interface.
I understand that the proper way is to order to my Telco two PRI lines
one for (S) and another for (A), and configure (S) and (A) to call
each other numbers when they have to interconnect.
Now, might it be possible to connect directly (A) and (S) using their
PSTN interfaces without having to go through to my Telco ?! Does it
make sense ? Is it technically feasible ? I guess that the Telco
network is providing routing, number assignation, etc. and it sounds
pointless to do this. Nevertheless could you confirm it is
possible/impossible and why ? Is there a better way to do that ?
Thanks in advance,
Pascal
------------------------------------------------------------------------
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stotaro at totarotechn... Guest
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Posted: Fri May 16, 2008 10:23 am Post subject: [asterisk-users] Connecting a PSTN gateway to Asterisk using |
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There is nothing special (beyond the regular configs) that need to be
done with Digium (Sangoma) or any compatible board. Since you are
using point to point data lines, I would suggest using SIP and
whatever codec fits your needs and bandwidth.
The rest is done it the dialplan.
Thanks,
Steve Totaro
On Fri, May 16, 2008 at 10:48 AM, Al Baker <bwentdg at pipeline.com> wrote:
Quote: | This is 'basically' a tie-line between the boxes.
Yes - it is done all the time between PBX's. You are basically nailing
up a circut between the boxes.
It could be a simple as a simple POTS leased line or a multi-t1 bundle
between them.
How it is physically done with DIGIUM's boards under * ?
Someone else will have to answer that
Pascal Maugeri wrote:
Quote: | Hi
I have a system (S) that has a PSTN gateway to accept incoming calls
and setup outgoing calls from/to Telco network. In the other hand I
have a distant Asterisk box (A) that I would like to connect to (S)
using the PRI interface.
I understand that the proper way is to order to my Telco two PRI lines
one for (S) and another for (A), and configure (S) and (A) to call
each other numbers when they have to interconnect.
Now, might it be possible to connect directly (A) and (S) using their
PSTN interfaces without having to go through to my Telco ?! Does it
make sense ? Is it technically feasible ? I guess that the Telco
network is providing routing, number assignation, etc. and it sounds
pointless to do this. Nevertheless could you confirm it is
possible/impossible and why ? Is there a better way to do that ?
Thanks in advance,
Pascal
------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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peter at boku.net Guest
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Posted: Fri May 16, 2008 10:44 am Post subject: [asterisk-users] Connecting a PSTN gateway to Asterisk using |
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Using a T1/E1 ISDN interface it's somewhat trivial. In zapata's conf:
group=0,11
context=from-pbx-custom
switchtype = national
signalling = pri_net
pridialplan=national
prilocaldialplan=national
channel => 1-23
group=
context=default
Note the pri_net for signalling. I have several PRI spans running this way
to PBXs. Then configure your dialplan to match the remote site's extensions
to use the right trunk interface.
On 5/16/08 9:48 AM, "Al Baker" <bwentdg at pipeline.com> wrote:
Quote: | This is 'basically' a tie-line between the boxes.
Yes - it is done all the time between PBX's. You are basically nailing
up a circut between the boxes.
It could be a simple as a simple POTS leased line or a multi-t1 bundle
between them.
How it is physically done with DIGIUM's boards under * ?
Someone else will have to answer that
Pascal Maugeri wrote:
Quote: | Hi
I have a system (S) that has a PSTN gateway to accept incoming calls
and setup outgoing calls from/to Telco network. In the other hand I
have a distant Asterisk box (A) that I would like to connect to (S)
using the PRI interface.
I understand that the proper way is to order to my Telco two PRI lines
one for (S) and another for (A), and configure (S) and (A) to call
each other numbers when they have to interconnect.
Now, might it be possible to connect directly (A) and (S) using their
PSTN interfaces without having to go through to my Telco ?! Does it
make sense ? Is it technically feasible ? I guess that the Telco
network is providing routing, number assignation, etc. and it sounds
pointless to do this. Nevertheless could you confirm it is
possible/impossible and why ? Is there a better way to do that ?
Thanks in advance,
Pascal
------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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stotaro at totarotechn... Guest
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Posted: Fri May 16, 2008 11:10 am Post subject: [asterisk-users] Connecting a PSTN gateway to Asterisk using |
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On point to point data, you will just be sending the calls over SIP.
While theoretically, you could use a Digium or Sangoma card to
terminate your data T1, I would suggest a Cisco box. Using G729 or
GSM (or even Speex if you are cool you can push many more calls
through the circuit than a point to point tie or voice circuit.
Thanks,
Steve Totaro
On Fri, May 16, 2008 at 11:44 AM, Peter Eisch <peter at boku.net> wrote:
Quote: |
Using a T1/E1 ISDN interface it's somewhat trivial. In zapata's conf:
group=0,11
context=from-pbx-custom
switchtype = national
signalling = pri_net
pridialplan=national
prilocaldialplan=national
channel => 1-23
group=
context=default
Note the pri_net for signalling. I have several PRI spans running this way
to PBXs. Then configure your dialplan to match the remote site's extensions
to use the right trunk interface.
On 5/16/08 9:48 AM, "Al Baker" <bwentdg at pipeline.com> wrote:
Quote: | This is 'basically' a tie-line between the boxes.
Yes - it is done all the time between PBX's. You are basically nailing
up a circut between the boxes.
It could be a simple as a simple POTS leased line or a multi-t1 bundle
between them.
How it is physically done with DIGIUM's boards under * ?
Someone else will have to answer that
Pascal Maugeri wrote:
Quote: | Hi
I have a system (S) that has a PSTN gateway to accept incoming calls
and setup outgoing calls from/to Telco network. In the other hand I
have a distant Asterisk box (A) that I would like to connect to (S)
using the PRI interface.
I understand that the proper way is to order to my Telco two PRI lines
one for (S) and another for (A), and configure (S) and (A) to call
each other numbers when they have to interconnect.
Now, might it be possible to connect directly (A) and (S) using their
PSTN interfaces without having to go through to my Telco ?! Does it
make sense ? Is it technically feasible ? I guess that the Telco
network is providing routing, number assignation, etc. and it sounds
pointless to do this. Nevertheless could you confirm it is
possible/impossible and why ? Is there a better way to do that ?
Thanks in advance,
Pascal
------------------------------------------------------------------------
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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