On Thu, Apr 27, 2017, at 09:10 AM, Artem Chekulaev 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
Both the chan_sip and chan_pjsip modules have a "media_address" option which can be used to specify the address to place in the SDP for media. In the case of chan_pjsip there is also a "bind_rtp_to_media_address" option which can be used to guarantee that RTP leaves from that same address as well. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- 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
