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

Thu Oct 29 19:42:44 CDT 2020

Hello,

Does anyone know a way with chan_sip to tell Asterisk to use a specific IP
address for its end of the communication for a specific device? Something
like:

[device]
type = friend
host = 11.22.11.22
ouraddress = 33.44.33.44

This is for use on a server with multiple IP addresses. There is the
"extenip" setting, but it's really designed for NAT, and can only appear in
the [general] section.

Any suggestions would be greatly appreciated.

On Sat, 24 Oct 2020 at 09:43, David Cunningham <dcunningham at voisonics.com>
wrote:

> 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
>

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