Hi All,
tl;dr - Kamailio is sending traffic to the Voip server with a source IP
of the private (vpn) interface.
For some weeks, I've been experimenting with Kamailio v 4.4.4, trying
different options, including running (or not) a vpn interface on
the kamailio server with and without NAT/rtpproxy. I have not as yet
fully "installed" kamailo with database etc, I have just been playing
with the config file, mostly running with -D
For most of the configurations I tried, I was routing SIP to/from the
VPN via another box on the network, and I tried to get RTP streams
running without using rtpproxy, sometimes it worked, but I have two many
double/triple NAT problems with some of the freeswitch nodes.
Now I have decided to run a VPN interface directly on the kamailio
server, for simplicity.
Essentially what I want to do is have kamailio/rtpproxy bridge between
the VPN and the VOIP provider, so it should be a pretty standard NAT config.
My setup looks like this:
VPN CLUSTER of Freeswitch nodes <-10.23.0.1-> KAMAILIO Server
<-192.168.1.1 - gateway - PUBLIC_IP-> Voip Provider
The Kamailio Server has two interfaces, one VPN interface and one facing
the Public Internet.
It is behind a firewall, and so has a private IP address assigned on the
interface, but the is dedicated public IP that forwards all relevant
ports, and traffic seen on the public side originates from this address.
I have two listen directives,
listen=udp:192.168.1.1:5060 advertise PUBLIC_IP:5060
listen=udp:10.23.0.1:5060
I expected Kamailio to receive SIP INVITE on the 10.23.0.1 address and
forward it to the voip provider via the default route (the 192.168.1.1
address), but it is sending the packet out to the voip provider with a
source address of 10.23.0.1 !!
Can anyone tell me how I can configure this? I cannot find anything
obvious in the manual.
Thanks!
Hi everybody,
I am installing siremis 4.3.0 with kamilio 5.1.0-dev1 but when i step to
the second step i get an error:
ERROR: SQLSTATE[23000]: Integrity constraint violation: 1062 Duplicate
entry '127.0.0.1' for key 'domain_idx'
Some solution?
Hello Guys,
I have a kamailio 4.2.8 receiving on tls and i'm trying to forward on tcp,
but AFTER the call is established, kamailio hangs the call with "SIPS
required"...
Has this happened to anyone?
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
ᐧ
Hello,
We have a use case where we suspend a transaction in a failure_route to give UEs that might be woken by a push notification more time to REGISTER and join the INVITE.
We'd like to drop the previous branches in this case. I tried using t_drop_replies() but it has no effect.
The doc states that t_drop_replies() is only working if a new branch is added. And from my understanding t_suspend() adds a new branch.
But is it possible that t_drop_replies() cannot be used with t_suspend()? Or am I missing something?
Kind Regards,
Andreas
Hi,
Thank you very much for the answers! I'll have a play.
Cheers,
Yufei
Message: 9
Date: Mon, 8 May 2017 01:01:51 -0700
From: Maxim Sobolev <sobomax(a)sippysoft.com>
To: sr-users(a)lists.kamailio.org, Daniel-Constantin Mierla
<miconda(a)gmail.com>
Subject: Re: [SR-Users] Can rtpproxy stream audio inside a call?
Message-ID:
<CAH7qZfs0tQTwa-aCwvHHhAS_hajCRNaKBLnjXBv+O0txJNd4Mw(a)mail.gmail.com>
Content-Type: text/plain; charset="utf-8"
Daniel is right. Technically speaking it can be done at any point after the
media session has been established, but you need some kind of trigger to
start/stop it from your routing script. Such as re-INVITE for example.
-Max
On May 8, 2017 7:52 AM, "Daniel-Constantin Mierla" <miconda(a)gmail.com>
wrote:
> Hello,
>
> it can play rtp files encoded in the format expected by the audio codec. I
> used only when there was a re-invite (like putting the call on hold).
>
> Cheers,
> Daniel
>
> On 04.05.17 17:29, Yufei Tao wrote:
>
> Hi,
>
> Just a quick question: if I use the Kamailio rtpproxy module with the
Sippy
> RTPproxy, can I stream audio within a call? Or is it only possible before
> call audio is established?
>
> For example can I mix/playback audio file into an existing established
> call session at any time using functions rtpproxy_stream2uac() and
rtpproxy_stream2uas()
> somehow? If it is possible, does it need to be triggered by an in-dialog
> message from either UAC or UAS?
>
> Cheers,
> Yufei
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing Listsr-users@lists.kamailio.orghttps://
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>
>
> --
> Daniel-Constantin Mierlawww.twitter.com/miconda
<http://mierlawww.twitter.com/miconda> -- www.linkedin.com/in/miconda
> Kamailio Advanced Training - May 22-24 (USA) - www.asipto.com
> Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
>
>
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>
>
Dear Daniel,
I am working on Push To Talk PoC. I intent to use Kamailio as SIP signaling
to work with SEMS for handling mixer conference audio (media server). The
reason to use SEMS is performance is better in comparing with FreeSwitch or
Asterisk. There are some guide to config Kamailio with Freeswitch or
Asterisk but no for SEMS.
If you have any information or guide line, please help!
Hi,
I discovered that when kamailio (version 5.0.1) starts, if reports a waring
on a 'if' command :
May 11 11:26:20 vm-vse02-siprouter2 kamailio: WARNING: <core>
[core/cfg.y:3378]: warn_at(): warning in config file
//etc/kamailio/kamailio.cfg, line 978, column 6-13: constant value in
if(...)
The line 978 (in yellow) is part of the following routing below in my
Kamailio.cfg :
route[TO_MMM] {
$var(mbxDN)=$null; # directory number containing mailbox
number
$var(uDigit)=$null; # application digits
$var(i:NodePath)=$null;
$var(TechNumber)=$null;
$avp(nodeInx)=$null;
# strip prefix from user in URI request user field
$var(TechNumber) =
$(rU{re.subst,/^(\+33|\+262|\+377|\+590|\+594|\+596|0)//});
# strip suffix from user in URI request user field - suffix
headed by anything other than digit or '+'
$var(TechNumber) =
$(var(TechNumber){re.subst,/[^\+0-9].*//});
if (VM_TARGET_TECHNICAL_NUMBER!=$null) {
$avp(vm_prefix) =
VM_TARGET_TECHNICAL_NUMBER;
} else {
$avp(vm_prefix) =
$(var(TechNumber){s.substr,0,5});
The same code with kamailio 3.3 doesn't report this warning. Also the same
code exists in another part of the routine but don't report a warning.
I have no idea of the impact of such warning.
Also I'm looking for help to understand the problem.
Thanks in advance for the answers.
Cordialement
Patrick GINHOUX
We have 1 stateless dispatcher and 3 proxy behind that dispatcher,
when Proxy send BYE message then dispatcher bouncing back that BYE to
Proxy and they playing ping pong. I think i am missing something..
How BYE get routed? in dispatcher i am not using loose_route()
function also should i need to use that but again we running
dispatcher in stateless.
Here i put SIP trace: https://pastebin.com/vp9JCdNP
Dispatcher config, Very simple..
route{
remove_hf("Route");
if (method == "INVITE") {
record_route();
}
if ( !ds_select_dst("1", "0") ) {
xlog("L_ERR", "Unable to route \n");
sl_send_reply("500","Unable to route");
break;
}
forward(uri:host, uri:port);
Hi,
I want to create conference without interact with IVR, so, trying to use
Conference Factory URI. https://tools.ietf.org/html/rfc4579#section-3.2
I am looking for documentation but it seems very poor documentation. Could
you please help to give more document or opensource for Conference Factory
URI. SEMS does not support Conference Factory URI I guess?
Thanks,
Nhan
