<div dir="ltr"><div>Hi Łukasz, <br></div><div><br></div><div>A TCP call works fine under normal circumstances. It's just when we send the call via a proxy that we have a problem, because the call to the proxy doesn't appear to use TCP.</div><div><br></div><div>Thank you.</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, 22 Jul 2022 at 11:58, Łukasz Grzywański <<a href="mailto:lukasz.grzywanski@ccig.pl">lukasz.grzywanski@ccig.pl</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr">Hi,<div>which version are you using ? <br></div><div>please show: asterisk -rx "sip show peer sip-peer"</div><div><br></div><div>I checked...</div><div>I use UDP and TCP, my phone via UDP, telekom via TCP and works </div><div><br></div><div><br></div><div>same => n,dial(SIP/${EXTEN}@sip-trunk-telekom)<br></div><div><br></div><div><img src="cid:ii_l5vnh4320" alt="image.png" style="margin-right: 0px;"><br></div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Thu, 21 Jul 2022 at 23:58, David Cunningham <<a href="mailto:dcunningham@voisonics.com" target="_blank">dcunningham@voisonics.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div>Thank you Thomas. I know it would be good to move to pjsip, and that's coming in a future product version, but it isn't used in the version of this scenario.</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, 22 Jul 2022 at 01:30, Thomas Ray <<a href="mailto:tom.ray@blazestudios.com" target="_blank">tom.ray@blazestudios.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex"><div lang="EN-CA"><div><p class="MsoNormal">The answer is chan_pjsip. You can do this with chan_pjsip. There’s no real support for chan_sip anymore. It’s dead, it’s going away. No fixes or updates will be accepted against it as of this point.<u></u><u></u></p><p class="MsoNormal"><u></u> <u></u></p><div style="border-color:rgb(181,196,223) currentcolor currentcolor;border-style:solid none none;border-width:1pt medium medium;padding:3pt 0cm 0cm"><p class="MsoNormal"><b><span style="font-size:12pt;color:black">From: </span></b><span style="font-size:12pt;color:black">asterisk-users <<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank">asterisk-users-bounces@lists.digium.com</a>> on behalf of Dovid Bender <<a href="mailto:dovid@telecurve.com" target="_blank">dovid@telecurve.com</a>><br><b>Reply-To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br><b>Date: </b>Thursday, July 21, 2022 at 9:21 AM<br><b>To: </b>Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com" target="_blank">asterisk-users@lists.digium.com</a>><br><b>Subject: </b>Re: [asterisk-users] TCP dial via proxy<u></u><u></u></span></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">David,<u></u><u></u></p><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">We had this exact "issue" in the past and were not able to figure out how to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp". So:<u></u><u></u></p></div><div><p class="MsoNormal">Dial(SIP/<a href="http://1234@1.1.1.1/2.2.2.2" target="_blank">1234@1.1.1.1//2.2.2.2</a>)<u></u><u></u></p></div><div><p class="MsoNormal">became:<u></u><u></u></p></div><div><div><p class="MsoNormal">Dial(SIP/<a href="http://force_tcp1234@1.1.1.1/2.2.2.2" target="_blank">force_tcp1234@1.1.1.1//2.2.2.2</a>)<u></u><u></u></p></div></div><div><p class="MsoNormal">On Kamailio's side in the FORWARD block we added:<u></u><u></u></p></div><div><p class="MsoNormal"># HACK for forcing TCP<br> if ($oU != $null && $(oU{s.len}) != 0) {<br> $var(prefix) = $(oU{s.substr,0,9});<br> if ($var(prefix) == "force_tcp") {<br> $rU = $(oU{s.substr,9,0});<br> add_uri_param( "transport=tcp" );<br> $fs = "tcp:" + $Ri + ":5060";<br> }<br> }<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div></div><p class="MsoNormal"><u></u> <u></u></p><div><div><p class="MsoNormal">On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <<a href="mailto:dcunningham@voisonics.com" target="_blank">dcunningham@voisonics.com</a>> wrote:<u></u><u></u></p></div><blockquote style="border-color:currentcolor currentcolor currentcolor rgb(204,204,204);border-style:none none none solid;border-width:medium medium medium 1pt;padding:0cm 0cm 0cm 6pt;margin-left:4.8pt;margin-right:0cm"><div><div><p class="MsoNormal">Hello,<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">We have an Asterisk dial which sends the call via a proxy using //, for example:<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Dial(SIP/${EXTEN}@peer_address//proxy_address)<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Does anyone know how we can make the SIP to the proxy use TCP? We tried making proxy_address match a peer in sip.conf with "transport = tcp" but that didn't seem to work. We are using chan_sip.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><div><p class="MsoNormal">Thanks very much for any advice.<u></u><u></u></p></div><div><p class="MsoNormal"><u></u> <u></u></p></div><p class="MsoNormal">-- <u></u><u></u></p><div><div><div><div><div><div><div><div><div><div><div><p class="MsoNormal">David Cunningham, Voisonics Limited<br><a href="http://voisonics.com/" target="_blank">http://voisonics.com/</a><br>USA: +1 213 221 1092<br>New Zealand: +64 (0)28 2558 3782<u></u><u></u></p></div></div></div></div></div></div></div></div></div></div></div></div><p class="MsoNormal">-- <br>_____________________________________________________________________<br>-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br><br>Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a><br><br>New to Asterisk? Start here:<br> <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/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" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><u></u><u></u></p></blockquote></div><p class="MsoNormal">-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> -- Check out the new Asterisk community forum at: <a href="https://community.asterisk.org/" target="_blank">https://community.asterisk.org/</a> New to Asterisk? Start here: <a href="https://wiki.asterisk.org/wiki/display/AST/Getting+Started" target="_blank">https://wiki.asterisk.org/wiki/display/AST/Getting+Started</a> asterisk-users mailing list To UNSUBSCRIBE or update options visit: <a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><u></u><u></u></p></div></div>
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<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/display/AST/Getting+Started</a><br>
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