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
