<div dir="ltr">Seems I responded the same time as Josh. Follow what he has suggested.<div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Thu, Apr 27, 2017 at 8:41 AM, Artem Chekulaev <span dir="ltr"><<a href="mailto:slonikk@gmail.com" target="_blank">slonikk@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">Yes, Voice = RTP</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">Using chan_sip</div></div><div class="gmail_extra"><br><div class="gmail_quote">2017-04-27 15:32 GMT+03:00 Dovid Bender <span dir="ltr"><<a href="mailto:dovid@telecurve.com" target="_blank">dovid@telecurve.com</a>></span>:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">By voice do you mean RTP? Are you using chan_sip or pjsip?<div><br></div></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div class="m_-5369961841037245729h5">On Thu, Apr 27, 2017 at 8:10 AM, Artem Chekulaev <span dir="ltr"><<a href="mailto:slonikk@gmail.com" target="_blank">slonikk@gmail.com</a>></span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="m_-5369961841037245729h5"><div dir="ltr"><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">I have connection with two networks (by VoIP provider setup)</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">1 - <a href="http://10.10.10.0/24" target="_blank">10.10.10.0/24</a> = SIP</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">2 - <a href="http://10.10.11.0/24" target="_blank">10.10.11.0/24</a> = Voice</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">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</div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small"><br></div><div class="gmail_default" style="font-family:verdana,sans-serif;font-size:small">Unfortunately, I _need_ to use two networks instead of one</div></div>
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