Hi Łukasz, 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.
Thank you. On Fri, 22 Jul 2022 at 11:58, Łukasz Grzywański <[email protected]> wrote: > Hi, > which version are you using ? > please show: asterisk -rx "sip show peer sip-peer" > > I checked... > I use UDP and TCP, my phone via UDP, telekom via TCP and works > > > same => n,dial(SIP/${EXTEN}@sip-trunk-telekom) > > [image: image.png] > > > On Thu, 21 Jul 2022 at 23:58, David Cunningham <[email protected]> > wrote: > >> Thank you Thomas. I know it would be good to move to pjsip, and that's >> coming in a future product version, but it isn't used in the version of >> this scenario. >> >> >> On Fri, 22 Jul 2022 at 01:30, Thomas Ray <[email protected]> >> wrote: >> >>> The answer is chan_pjsip. You can do this with chan_pjsip. There’s no >>> real support for chan_sip anymore. It’s dead, it’s going away. No fixes or >>> updates will be accepted against it as of this point. >>> >>> >>> >>> *From: *asterisk-users <[email protected]> on >>> behalf of Dovid Bender <[email protected]> >>> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion < >>> [email protected]> >>> *Date: *Thursday, July 21, 2022 at 9:21 AM >>> *To: *Asterisk Users Mailing List - Non-Commercial Discussion < >>> [email protected]> >>> *Subject: *Re: [asterisk-users] TCP dial via proxy >>> >>> >>> >>> David, >>> >>> >>> >>> We had this exact "issue" in the past and were not able to figure out >>> how to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp". >>> So: >>> >>> Dial(SIP/[email protected]//2.2.2.2 <http://[email protected]/2.2.2.2>) >>> >>> became: >>> >>> Dial(SIP/[email protected]//2.2.2.2 >>> <http://[email protected]/2.2.2.2>) >>> >>> On Kamailio's side in the FORWARD block we added: >>> >>> # HACK for forcing TCP >>> if ($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 >> >> >> >> -- >> 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 > > > > -- > > Pozdrawiam, > > Łukasz Grzywański > Voice Architect > > Mok Yok IT Sp. z o.o. > ul. Rzeźnicza 32/33, 50-130 Wrocław, Polska > tel. +48 717227200, fax +48 717227299 > mob.: +48 517255333, e-mail: [email protected] > -- > _____________________________________________________________________ > -- 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
