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.
On Fri, 22 Jul 2022 at 01:30, Thomas Ray <tom....@blazestudios.com> wrote: > 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 <asterisk-users-boun...@lists.digium.com> on > behalf of Dovid Bender <do...@telecurve.com> > *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *Date: *Thursday, July 21, 2022 at 9:21 AM > *To: *Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > *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/1234@1.1.1.1//2.2.2.2 <http://1234@1.1.1.1/2.2.2.2>) > > became: > > Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2 > <http://force_tcp1234@1.1.1.1/2.2.2.2>) > > 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:" + $Ri + ":5060"; > } > } > > > > > > > > On Wed, Jul 20, 2022 at 10:47 PM David Cunningham < > dcunning...@voisonics.com> wrote: > > Hello, > > > > We have an Asterisk dial which sends the call via a proxy using //, for > example: > > > > Dial(SIP/${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 -- 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