[asterisk-users] Multiple IP addresses and using same IP for outbound calls as inbound
John Runyon
john at simplynuc.com
Thu Oct 29 22:23:40 CDT 2020
David, can you play around with the routing table and get the OS to handle
it for you? So long as asterisk isn’t calling bind() (or is calling with
0.0.0.0) I would imagine adding a route for the peer, with your normal
gateway, and the correct device would work.
On Thu, Oct 29, 2020 at 10:04 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
--
*John Runyon* | SimplyNUC <https://simplynuc.com> | Network Administrator
O: (512) 766-0401 x1110
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