If you are using PJSIP, you should be able to define a different transport for each source IP that you want to use and simply tie the endpoint to your provider to the appropriate transport. Obviously, you'd need to define the IP on the interface as well. You could use the same interface with multiple IPs or, if you need to separate the traffic over unique links (not likely, I imagine), you could always tie them to different interfaces if needed.
On Fri, May 27, 2016 at 1:09 AM Attila Megyeri <amegy...@minerva-soft.com> wrote: > > If I had just two such users, I would be fine. > But having 10-20 or even more? It is not a nice, scalable solution... > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] On Behalf Of Neeraj Chand > Sent: Thursday, May 26, 2016 11:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > Subject: Re: [asterisk-users] Sending Calls via SIP trunk from several > different IP addresses from same Asterisk Machine, to the same destination > IP > > How about running a second asterisk instance on the same box with > different IP/Port combo > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users