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 <dcunning...@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 <dcunning...@voisonics.com> > wrote: > >> 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 >> > > > -- > 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