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

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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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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Why not use OpenSips/Kamailoo in between? Where you want 1.1.1.1 you pass
it along as is. Where you want 2.2.2.2 change the sdp in opensips/kamailio

On Thu, Oct 29, 2020 at 20:44 David Cunningham <dcunningham at
voisonics.com>
wrote:
> 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
> --
> _____________________________________________________________________
> -- 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
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