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

[George Joseph]

On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <dcunningham at voisonics.com>
wrote:

> 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.
>

Sorry.  I just assume chan_pjsip these days.  Not sure how you'd do it for
chan_sip.



>
>
> 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
> --
> _____________________________________________________________________
> -- 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]
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201023/4a8c4b49/attachment.html>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: image.png
Type: image/png
Size: 5142 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201023/4a8c4b49/attachment.png>