asterisk users - Oct 2020 - Multiple IP addresses and using same IP for outbound calls as inbound

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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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]
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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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