Hi Dovid,
Thanks for the reply. We are indeed able to force TCP from the Kamailio
proxy, but haven't been able to force it between Asterisk and Kamailio.
On Fri, 22 Jul 2022 at 01:21, Dovid Bender <do...@telecurve.com> wrote:
> 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
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> Check out the new Asterisk community forum at:
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>>
>> New to Asterisk? Start here:
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>>
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>
> --
> _____________________________________________________________________
> -- 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:
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