[SR-Users] Contents of sr-users Digest, Vol 96, Issue 68
Jignesh Gandhi
Jignesh.Gandhi at MoviusCorp.com
Mon May 20 21:32:34 CEST 2013
Thanks for the quick reply.
1. Re: SCTP question (Daniel-Constantin Mierla)
What I meant is if the SCTP association goes down from the Invite, 200 ok w/sdp to BYE.
Should there be not an error at t_relay() level , since the association does not exists ?
Currently I don't get any error when I do
if (!t_relay()) {
xlog("L_INFO", "T_Relay return code is $retcode\n");
sl_reply_error();
}
I have set up $du with the IP:port of the host where I received the Invite and ACK to 200 ok w/sdp.
I have shut down the other side , so I know that SCTP is not up.
Thanks,
--Jignesh
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Today's Topics:
1. Re: SCTP question (Daniel-Constantin Mierla)
2. Re: [SR-Users]
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
(Daniel-Constantin Mierla)
3. Re: Kamailio + Siremis Outbound route (Daniel-Constantin Mierla)
4. SCTP support of MultiHoming... (Jignesh Gandhi)
5. Re: SCTP support of MultiHoming... (Daniel-Constantin Mierla)
----------------------------------------------------------------------
Message: 1
Date: Mon, 20 May 2013 16:56:55 +0200
From: Daniel-Constantin Mierla <miconda at gmail.com>
To: Jignesh Gandhi <jigpgandhi at gmail.com>, "Kamailio (SER) - Users
Mailing List" <sr-users at lists.sip-router.org>
Subject: Re: [SR-Users] SCTP question
Message-ID: <519A39B7.3050503 at gmail.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
Hello,
On 5/20/13 6:12 AM, Jignesh Gandhi wrote:
> Hello,
>
> I recently started using SCTP relay feature of Kamailio and am
> receiving SCTP-SIP and relaying it UDP-SIP and vice versa.
>
> What happens , if an SCTP association is broken from the distant end
> in the middle of the call, is there a way to re transmit the SCTP
> message via another route ?
By middle of the call, do you mean in between the INVITE and the BYE? I am not that familiar with SCTP, but probably a new one is created with the BYE. The same should be happening with tcp.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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Message: 2
Date: Mon, 20 May 2013 17:13:22 +0200
From: Daniel-Constantin Mierla <miconda at gmail.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users at lists.sip-router.org>
Subject: Re: [SR-Users]
http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour
Message-ID: <519A3D92.7030501 at gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello,
On 5/20/13 12:47 PM, johnc wrote:
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi,
>
> I updated this config to work with version 4. See below. I have calls
> working over TLS between jitsi sip clients registered to the same
> proxy of either one of two proxies I have built using this updated
> config. I can make calls to/from clients over TLS registered to the
> openrcs.com proxy from both of these proxies. I can't make calls
> between the two proxies configured with the config below. Each domain
> has commercial SSL certs. I have rtpproxy configured and working. I
> would be very grateful if somebody would check the config and see if I have made a mistake.
> Many thanks. I will post the final working config so it may be of help
> to others.
it is rather impossible to test configs from other people and not easy to review large configs. So I recommend you run with debug=3, see what is printed when you try a call that fails. If you cannot figure out from there what's the solution, then send the debug messages here.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
------------------------------
Message: 3
Date: Mon, 20 May 2013 17:18:37 +0200
From: Daniel-Constantin Mierla <miconda at gmail.com>
To: tony.turner at nodemax.com, "Kamailio (SER) - Users Mailing List"
<sr-users at lists.sip-router.org>
Subject: Re: [SR-Users] Kamailio + Siremis Outbound route
Message-ID: <519A3ECD.30206 at gmail.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
Hello,
if you want to send all calls that arrive to kamailio having the prefix
01 to freeswitch:
if($rU =~"^01") {
$ru = "sip:" + $rU + "@__FREESWITCHIP__";
route(RELAY);
exit;
}
Be sure calls are authenticated at that point and, if needed, the call
is not actually coming from freeswitch.
Cheers,
Daniel
On 5/20/13 11:33 AM, Tony Turner wrote:
>
> Hi
>
> Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get
> install
>
> I want to use Kamailio as a proxy edge register to our network.
>
> I have installed Kamailio and freeswitch.
>
> I can register on Kamailio but I can't route a call from my sip client
> from Kamailio to freeswitch and out to PSTN
>
> Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway ---
> Carriers
>
> If I register direct on Freeswitch I can route out to PSTN but I don't
> understand Kamailio routing.
>
> Can someone let me how I route say from SIP client registered on
> Kamailio to prefix 01% which goes out to Freeswitch
>
> Many Thanks
>
> Tony
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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Message: 4
Date: Mon, 20 May 2013 13:06:00 -0400
From: Jignesh Gandhi <Jignesh.Gandhi at MoviusCorp.com>
To: "sr-users at lists.sip-router.org" <sr-users at lists.sip-router.org>
Subject: [SR-Users] SCTP support of MultiHoming...
Message-ID:
<B3EDF1230C0A264298C8FCA4A9C0A97F4FC6792377 at gemsex02.gems.glenayre.com>
Content-Type: text/plain; charset="us-ascii"
Hello,
I read in the documentation that Kamailio SCTP supports multi homing.
Does anyone know how I can configure this?
Thanks in advance,
--Jignesh Gandhi
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Message: 5
Date: Mon, 20 May 2013 20:24:34 +0200
From: Daniel-Constantin Mierla <miconda at gmail.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users at lists.sip-router.org>
Subject: Re: [SR-Users] SCTP support of MultiHoming...
Message-ID: <519A6A62.8000602 at gmail.com>
Content-Type: text/plain; charset="iso-8859-1"; Format="flowed"
Hello,
maybe the commit log helps a bit:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=commit;h=eb321e3ffeb7ff474693bc035c8af0915a745b4f
Cheers,
Daniel
On 5/20/13 7:06 PM, Jignesh Gandhi wrote:
>
> Hello,
>
> I read in the documentation that Kamailio SCTP supports multi homing.
>
> Does anyone know how I can configure this?
>
> Thanks in advance,
>
> --Jignesh Gandhi
>
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
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