[asterisk-users] TCP dial via proxy
David Cunningham
dcunningham at voisonics.com
Thu Jul 21 19:00:25 CDT 2022
Hi Łukasz,
A TCP call works fine under normal circumstances. It's just when we send
the call via a proxy that we have a problem, because the call to the proxy
doesn't appear to use TCP.
Thank you.
On Fri, 22 Jul 2022 at 11:58, Łukasz Grzywański <lukasz.grzywanski at ccig.pl>
wrote:
> Hi,
> which version are you using ?
> please show: asterisk -rx "sip show peer sip-peer"
>
> I checked...
> I use UDP and TCP, my phone via UDP, telekom via TCP and works
>
>
> same => n,dial(SIP/${EXTEN}@sip-trunk-telekom)
>
> [image: image.png]
>
>
> On Thu, 21 Jul 2022 at 23:58, David Cunningham <dcunningham at voisonics.com>
> wrote:
>
>> Thank you Thomas. I know it would be good to move to pjsip, and that's
>> coming in a future product version, but it isn't used in the version of
>> this scenario.
>>
>>
>> On Fri, 22 Jul 2022 at 01:30, Thomas Ray <tom.ray at blazestudios.com>
>> wrote:
>>
>>> The answer is chan_pjsip. You can do this with chan_pjsip. There’s no
>>> real support for chan_sip anymore. It’s dead, it’s going away. No fixes or
>>> updates will be accepted against it as of this point.
>>>
>>>
>>>
>>> *From: *asterisk-users <asterisk-users-bounces at lists.digium.com> on
>>> behalf of Dovid Bender <dovid at telecurve.com>
>>> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion <
>>> asterisk-users at lists.digium.com>
>>> *Date: *Thursday, July 21, 2022 at 9:21 AM
>>> *To: *Asterisk Users Mailing List - Non-Commercial Discussion <
>>> asterisk-users at lists.digium.com>
>>> *Subject: *Re: [asterisk-users] TCP dial via proxy
>>>
>>>
>>>
>>> David,
>>>
>>>
>>>
>>> We had this exact "issue" in the past and were not able to figure out
>>> how to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp".
>>> So:
>>>
>>> Dial(SIP/1234 at 1.1.1.1//2.2.2.2 <http://1234@1.1.1.1/2.2.2.2>)
>>>
>>> became:
>>>
>>> Dial(SIP/force_tcp1234 at 1.1.1.1//2.2.2.2
>>> <http://force_tcp1234@1.1.1.1/2.2.2.2>)
>>>
>>> On Kamailio's side in the FORWARD block we added:
>>>
>>> # HACK for forcing TCP
>>> if ($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:" + $Ri + ":5060";
>>> }
>>> }
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
>>> dcunningham at voisonics.com> wrote:
>>>
>>> Hello,
>>>
>>>
>>>
>>> We have an Asterisk dial which sends the call via a proxy using //, for
>>> example:
>>>
>>>
>>>
>>> Dial(SIP/${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
>>> --
>>> _____________________________________________________________________
>>> -- 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:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
>
> Pozdrawiam,
>
> Łukasz Grzywański
> Voice Architect
>
> Mok Yok IT Sp. z o.o.
> ul. Rzeźnicza 32/33, 50-130 Wrocław, Polska
> tel. +48 717227200, fax +48 717227299
> mob.: +48 517255333, e-mail: lukasz.grzywanski at mokyokit.com
> --
> _____________________________________________________________________
> -- 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
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