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/${EXTEN}@sip-trunk-telekom)
[image: image.png]
On Thu, 21 Jul 2022 at 23:58, David Cunningham <dcunning...@voisonics.com>
wrote:
> Thank you Thomas. I know it would be good to move to pjsip, and that's
> coming in a future product version, but it isn't used in the version of
> this scenario.
>
>
> On Fri, 22 Jul 2022 at 01:30, Thomas Ray <tom....@blazestudios.com> wrote:
>
>> The answer is chan_pjsip. You can do this with chan_pjsip. There’s no
>> real support for chan_sip anymore. It’s dead, it’s going away. No fixes or
>> updates will be accepted against it as of this point.
>>
>>
>>
>> *From: *asterisk-users <asterisk-users-boun...@lists.digium.com> on
>> behalf of Dovid Bender <do...@telecurve.com>
>> *Reply-To: *Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Date: *Thursday, July 21, 2022 at 9:21 AM
>> *To: *Asterisk Users Mailing List - Non-Commercial Discussion <
>> asterisk-users@lists.digium.com>
>> *Subject: *Re: [asterisk-users] TCP dial via proxy
>>
>>
>>
>> David,
>>
>>
>>
>> We had this exact "issue" in the past and were not able to figure out how
>> to do it. Where we wanted tcp we prefixed the sip URI with "force_tcp". So:
>>
>> Dial(SIP/1234@1.1.1.1//2.2.2.2 <http://1234@1.1.1.1/2.2.2.2>)
>>
>> became:
>>
>> Dial(SIP/force_tcp1234@1.1.1.1//2.2.2.2
>> <http://force_tcp1234@1.1.1.1/2.2.2.2>)
>>
>> On Kamailio's side in the FORWARD block we added:
>>
>> # HACK for forcing TCP
>> if ($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:" + $Ri + ":5060";
>> }
>> }
>>
>>
>>
>>
>>
>>
>>
>> On Wed, Jul 20, 2022 at 10:47 PM David Cunningham <
>> dcunning...@voisonics.com> wrote:
>>
>> Hello,
>>
>>
>>
>> We have an Asterisk dial which sends the call via a proxy using //, for
>> example:
>>
>>
>>
>> Dial(SIP/${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:
>> 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.grzywan...@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
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