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

[David Cunningham]

OK, thank you George.


On Sat, 24 Oct 2020 at 03:16, George Joseph <gjoseph at digium.com> wrote:

>
>
> 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]
> --
> _____________________________________________________________________
> -- 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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