Seems I responded the same time as Josh. Follow what he has suggested.
On Thu, Apr 27, 2017 at 8:41 AM, Artem Chekulaev <slon...@gmail.com> wrote: > Yes, Voice = RTP > > Using chan_sip > > 2017-04-27 15:32 GMT+03:00 Dovid Bender <do...@telecurve.com>: > >> By voice do you mean RTP? Are you using chan_sip or pjsip? >> >> >> On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev <slon...@gmail.com> >> wrote: >> >>> I have connection with two networks (by VoIP provider setup) >>> 1 - 10.10.10.0/24 = SIP >>> 2 - 10.10.11.0/24 = Voice >>> >>> How to tell Asterisk send / receive voice traffic not on SIP network. >>> When I look into dumps, I see Asterisk trying to use SIP net for voice >>> >>> Unfortunately, I _need_ to use two networks instead of one >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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