On Thu, Oct 22, 2020 at 4:13 PM David Cunningham <dcunning...@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 <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 -- 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