?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? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170427/49046b3f/attachment.html>
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
By voice do you mean RTP? Are you using chan_sip or pjsip? On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev <slonikk at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170427/833e4f8c/attachment.html>
Yes, Voice = RTP Using chan_sip 2017-04-27 15:32 GMT+03:00 Dovid Bender <dovid at 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 <slonikk at 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20170427/d0ace45b/attachment.html>