OK, thank you George.
On Sat, 24 Oct 2020 at 03:16, George Joseph <gjos...@digium.com> wrote: > > > 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 -- 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