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
> --
> _____________________________________________________________________
> -- 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
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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
--
_____________________________________________________________________
-- 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
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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 <dovid at 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 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 >> -- >> _____________________________________________________________________ >> -- 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20220722/d75a2a92/attachment.html>