[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound
Dovid Bender
dovid at telecurve.com
Fri Oct 30 05:44:53 CDT 2020
Run rtp proxy on the asterisk box (not sure if it would work since you
can't use the same ports).
On Thu, Oct 29, 2020 at 11:03 PM David Cunningham <dcunningham at voisonics.com>
wrote:
> Hi Dovid,
>
> We can change the SDP in Kamailio, but Asterisk will still send its RTP
> from its default address. The remote end is strict about accepting RTP from
> the specified source and won't accept it. Have you any suggestions to solve
> that problem?
>
> Thank you.
>
>
> On Fri, 30 Oct 2020 at 14:49, Dovid Bender <dovid at telecurve.com> wrote:
>
>> 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
>>
>> --
>> _____________________________________________________________________
>> -- 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
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