<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>
<br></div></div><span>--<br>
______________________________<wbr>______________________________<wbr>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.org<wbr>/</a><br>
<br>
New to Asterisk? Start here:<br>
      <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki<wbr>/display/AST/Getting+Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailm<wbr>an/listinfo/asterisk-users</a><br></span></blockquote></div><br></div><span class="HOEnZb"><font color="#888888">
<br>--<br>
______________________________<wbr>______________________________<wbr>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.org<wbr>/</a><br>
<br>
New to Asterisk? Start here:<br>
      <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/wiki<wbr>/display/AST/Getting+Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/mailm<wbr>an/listinfo/asterisk-users</a><br></font></span></blockquote></div><br></div>
<br>--<br>
______________________________<wbr>______________________________<wbr>_________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" rel="noreferrer" target="_blank">http://www.api-digital.com</a> --<br>
<br>
Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" rel="noreferrer" target="_blank">https://community.asterisk.<wbr>org/</a><br>
<br>
New to Asterisk? Start here:<br>
      <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" rel="noreferrer" target="_blank">https://wiki.asterisk.org/<wbr>wiki/display/AST/Getting+<wbr>Started</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
   <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" rel="noreferrer" target="_blank">http://lists.digium.com/<wbr>mailman/listinfo/asterisk-<wbr>users</a><br></blockquote></div><br></div>