David, can you play around with the routing table and get the OS to handle it for you? So long as asterisk isn’t calling bind() (or is calling with 0.0.0.0) I would imagine adding a route for the peer, with your normal gateway, and the correct device would work.
On Thu, Oct 29, 2020 at 10:04 PM David Cunningham <dcunning...@voisonics.com> wrote: > Hi Dovid, > > We can change the SDP in Kamailio, but Asterisk will still send its RTP > from its default address. The remote end is strict about accepting RTP from > the specified source and won't accept it. Have you any suggestions to solve > that problem? > > Thank you. > > > On Fri, 30 Oct 2020 at 14:49, Dovid Bender <do...@telecurve.com> wrote: > >> 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 > > > > -- > 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 -- *John Runyon* | SimplyNUC <https://simplynuc.com> | Network Administrator O: (512) 766-0401 x1110 [image: Simply NUC] The Value of Purchasing From Simply NUC <https://simplynuc.com/about/> GSA Contract 47QTCA19D006U 495 Round Rock West Drive Round Rock, TX 78681 <https://www.google.com/maps/place/495+Round+Rock+W+Dr,+Round+Rock,+TX+78681/> See our reviews on [image: Trustpilot] <https://www.trustpilot.com/review/simplynuc.com?utm_medium=Trustbox&utm_source=EmailSignature1>
-- _____________________________________________________________________ -- 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