[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound

David Cunningham dcunningham at voisonics.com
Thu Oct 22 17:11:41 CDT 2020


Hi George,

Thank you for the response. I'm a little unclear on what you mean by a
transport. We're using chan_sip, not pjsip.

Do you mean a device in sip.conf, using bindaddr to set the address to bind
for that device? We've only used bindaddr in the [general] section before,
but if it will work in a device that could be the answer.


On Fri, 23 Oct 2020 at 00:13, George Joseph <gjoseph at digium.com> wrote:

>
>
> On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <
> dcunningham at 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
>> destination at termination.com -> INVITE sent from 1.1.1.1:5060 to
>> termination.com
>> INVITE received to 2.2.2.2:5060 -> Asterisk dials destination at 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



-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782
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