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)
became:
Dial(SIP/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
> --
> _____________________________________________________________________
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_____________________________________________________________________
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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
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