[asterisk-users] TCP dial via proxy

[Thomas Ray]

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-bounces at lists.digium.com> on behalf of Dovid Bender <dovid at telecurve.com>
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com>
Date: Thursday, July 21, 2022 at 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at 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 at 1.1.1.1//2.2.2.2)

became:

Dial(SIP/force_tcp1234 at 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 <dcunningham at 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

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