Yes, Voice = RTP Using chan_sip
2017-04-27 15:32 GMT+03:00 Dovid Bender <[email protected]>: > By voice do you mean RTP? Are you using chan_sip or pjsip? > > > On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev <[email protected]> > 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
