On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <dcunning...@voisonics.com>
wrote:

> Hello,
>
> We have an Asterisk server with two public IP addresses, let's say 1.1.1.1
> and 2.2.2.2. Normally calls come in to 1.1.1.1 and are bridged with a call
> dialled from Asterisk to an external destination. The external destination
> sees the SIP packet as coming from 1.1.1.1 and the media address in the SDP
> is 1.1.1.1, which is great.
>
> However if we receive a call in to 2.2.2.2 then the call dialled from
> Asterisk to an external destination still comes from 1.1.1.1, whereas we
> want it to come from 2.2.2.2. The source of any dialled call (the IP packet
> and the SDP media address) should be the same as the address the related
> inbound call was received to.
>
> For example:
> INVITE received to 1.1.1.1:5060 -> Asterisk dials
> destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to
> termination.com
> INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@pstn.com ->
> INVITE sent from 2.2.2.2:5060 to pstn.com
>
> Does anyone know how this can be achieved?
>

If termination.com is only on 1.1.1.1 and pstn.com is only on 2.2.2.2,
create 2 transports, one specifically bound to 1.1.1.1, transport-1.1.1.1
for instance, and another to 2.2.2.2:  transport-2.2.2.2.  The names aren't
important as long as you can tell the difference.  Then explicitly
configure endpoint termination.com's "transport" parameter to
"transport-1.1.1.1" and pstn.com's "transport" parameter to
"transport-2.2.2.2".   In your dialplan, you can see which endpoint the
call came in on, and route it out the same endpoint.

If both providers are available from both interfaces, you can create 2
endpoint for each provider: termination.com-1.1.1.1, pstn.com-1.1.1.1,
termination.com-2.2.2.2 and pstn.com-2.2.2.2;  Then configure each with the
same transports as above.





>
> Thanks in advance for your help,
>
> --
> David Cunningham, Voisonics Limited
> http://voisonics.com/
> USA: +1 213 221 1092
> New Zealand: +64 (0)28 2558 3782
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users



-- 
George Joseph
Asterisk Software Developer
direct/fax +1 256 428 6012
Check us out at www.sangoma.com and www.asterisk.org
[image: image.png]
-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to