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 <gjos...@digium.com> wrote: > > > 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 -- 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