On Wed, Oct 21, 2020 at 9:16 PM David Cunningham <dcunning...@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 > destinat...@termination.com -> INVITE sent from 1.1.1.1:5060 to > termination.com > INVITE received to 2.2.2.2:5060 -> Asterisk dials destinat...@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
