Re: TCP dial via proxy

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
>
> --
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> Check out the new Asterisk community forum at:
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>
> New to Asterisk? Start here:
>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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-- 
David Cunningham, Voisonics Limited
http://voisonics.com/
USA: +1 213 221 1092
New Zealand: +64 (0)28 2558 3782

-- 
_____________________________________________________________________
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Check out the new Asterisk community forum at: https://community.asterisk.org/

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      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

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