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.
From: asterisk-users <[email protected]> on behalf of Dovid Bender <[email protected]> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Date: Thursday, July 21, 2022 at 9:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <[email protected]> Subject: Re: [asterisk-users] TCP dial via proxy David, 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: Dial(SIP/[email protected]//2.2.2.2) became: Dial(SIP/[email protected]//2.2.2.2) On Kamailio's side in the FORWARD block we added: # HACK for forcing TCP if ($oU != $null && $(oU{s.len}) != 0) { $var(prefix) = $(oU{s.substr,0,9}); if ($var(prefix) == "force_tcp") { $rU = $(oU{s.substr,9,0}); add_uri_param( "transport=tcp" ); $fs = "tcp:" + $Ri + ":5060"; } } On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <[email protected]> wrote: Hello, We have an Asterisk dial which sends the call via a proxy using //, for example: Dial(SIP/${EXTEN}@peer_address//proxy_address) 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. Thanks very much for any advice. -- David Cunningham, Voisonics Limited http://voisonics.com/ USA: +1 213 221 1092 New Zealand: +64 (0)28 2558 3782 -- _____________________________________________________________________ -- 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
-- _____________________________________________________________________ -- 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
