Hello,
have you set the advertise address with public for listen socket (on
private ip)?
Cheers,
Daniel
On 06/02/2017 21:25, JBF wrote:
> Hello, yes, we kept TLS between kamailio and F5, and kamailio indeed have a
> private address, and doesnt see the firewall public address: the only way
> for
Hello, yes, we kept TLS between kamailio and F5, and kamailio indeed have a
private address, and doesnt see the firewall public address: the only way
for the proxy to reply to the F5 is through the client socket opened by
the F5 connection
--
View this message in context:
http://sip-router.10
Hello,
is Kamailio also listening on TLS? Or is the firewall converting to UDP
or TCP?
Has Kamailio a private IP and only the firewall a public IP?
Cheers,
Daniel
On 25/01/2017 17:23, JBF wrote:
> Hello,
> we have the following Configuration for our kamailio installation (we are
> using TLS a
Hello,
we have the following Configuration for our kamailio installation (we are
using TLS and not udp)
(1) F5 Firewall (configured as message fowarding), opening a TLS server on
the outside
(2) SIP proxy, with a TLS server accessed by the F5 . The SIP proxy doesnt
see the F5 TLS server
(3) SIP r
What kind of suggestions are you looking for?
If you're looking for validation of your intentions: yes, Kamailio can
be used for this purpose. Indeed, it is one of the more common use-cases.
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 303
hi ，
I want to use kamailio as a proxy ;
/
gateway1 ->pstn
/
1、 call from backend server ---> kamailio ->gateway2 -> pstn
hi ，
I want to use kamailio as a proxy ;
/
gateway1 ->pstn
/
1、 call from backend server ---> kamailio ->gateway2 -> pstn
I got bridging working well on internal interfaces in case of simple SIP
calls on a bit other configuration. But editing this config to support
WebRTC causes same problems. I need internal interfaces on asterisk to
completely close external ones (Security etc.).
Do you have proper routing rules between the local ips of kamailio and
asterisk? Why aren't you use only external IPs if they are on different
servers? Asterisk has also the option to set external ip. It can reduce
the complexity of doing bridging of signaling and rtp. Once you get that
working you
Asterisk localip=10.0.0.87, sorry
2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Ok, so my scheme.
> Kamailio and Asterisk are in Amazon EC2
> Kamailio externip=54.197.230.121 localip=10.145.45.103
> Asterisk localip=10.145.45.103, externip doesn't matter
>
> Call should flo
Ok, so my scheme.
Kamailio and Asterisk are in Amazon EC2
Kamailio externip=54.197.230.121 localip=10.145.45.103
Asterisk localip=10.145.45.103, externip doesn't matter
Call should flow like that:
webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip
but now it's webrtc --> kam
Also, an interesting thing - if you can see in Kamailio log, a check of the
proto of user "300" is being made. But 300 is $tU, and $tU proto is being
checked only if source IP is asterisks IP.
Here's the part of config where rtpengine is engaged (in NATmanage route)
if((src_ip==10.0.0.87)
Can you specify exactly which side received what IP and what you would
expect there? It is not easy to digests lots of logs and also guess what
would you expect to happen...
Cheers,
Daniel
On 24/06/15 15:14, Alexandru Covalschi wrote:
> Heh...
> Well, I still have troubles with my configuration.
Heh...
Well, I still have troubles with my configuration. And in SDP media adress
is Amazon public interface - but rtpengine has replace-origin
replace-session-connection session, so it must be local address.
Any ideas?
Asterisk log http://pastebin.com/MFt9V9qK
Kamailio log http://pastebin.com/jZce
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
> works on other then Amazon EC2 environment and I still get this error.
> Maybe it is somehow
I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
works on other then Amazon EC2 environment and I still get this error.
Maybe it is somehow related to NAT traversal?
Kamailio log: http://pastebin.com/jZceP2Rn
javascript log: http://pastebin.com/9Y4Pv43W
2015-06-23 20:40 G
Here is it
http://pastebin.com/JkkM4M5m
2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla :
> There are no major changes in 4.3 comparing with 4.2 in regards to
> websocket -- the implementation is quite mature for a long time.
>
> Looks like websocket connection is not available. Can you look
There are no major changes in 4.3 comparing with 4.2 in regards to
websocket -- the implementation is quite mature for a long time.
Looks like websocket connection is not available. Can you look at
javascript debug console in the browser to see what is printing?
Daniel
On 23/06/15 17:23, Alexand
without fix_nated_contact error behaviour is the same
maybe I should upgrade to 4.3 ?
2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Here's the trace on port which I use for ws server. Don't look at
> fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
Here's the trace on port which I use for ws server. Don't look at
fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
establish a ws connection properly. Client is SIPML5 demo phone
http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alexandru Covalschi <568...@gmail.com
I solved the SIP voice trouble, but WebRTC problem still exists. What kind
of trace I must do to make my post more informative?
2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla :
> Hello,
>
> On 23/06/15 04:10, Alexandru Covalschi wrote:
>
> Hello. I'm trying to set up this (v 4.2 stable):
>
Hello,
On 23/06/15 04:10, Alexandru Covalschi wrote:
> Hello. I'm trying to set up this (v 4.2 stable):
> peer <--> ec2 <--kamailio+rtpengine--> asterisk
> scheme
>
> I use advertised adress for SIP and WS connections.
> The problem is that on SIP I get one way audio - I can receive audio
> from a
Hello. I'm trying to set up this (v 4.2 stable):
peer <--> ec2 <--kamailio+rtpengine--> asterisk
scheme
I use advertised adress for SIP and WS connections.
The problem is that on SIP I get one way audio - I can receive audio from
asterisk, but I can't transmit audio there - my SIP UA tries to send
Hello Daniel, Vasiliy
Did you get a chance to look at it ?
I tried the same experiment with CSipSimple + Video plugin but results are
unfortunately the same :(
So does it comes down to the configuration of the proxy ? Or something else
?
Really need some guidance here.
Thanks!
On Wed, May 20,
Hello Vasiliy,
Thanks for replying.
Not sure why Linphone-Android won't receive INVITE, since it is responding
well to the keep-alive OPTIONS messages from proxy.
Both, proxy and SIP server are sending packets to UAb on which UAb is
apparently responding to only proxy.
Is this a genuine flaw in
Looks like UAb (3G) do not receive the INVITE (or makes no answer for some
reason).
Can you check if UAb is receiving the INVITE?
LinphoneAndroid that you use as a User-Agent on UAb has to log something.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Kamailio-proxy-f
have to tune the nat_uac_test() parameter in order to match
> what you want to relay.
>
> For signaling relaying, be sure you do record_route() for calls.
>
> Cheers,
> Daniel
>
>
> On 07/05/15 16:01, rahul.ultimate wrote:
>
> Hi Daniel
>
> I have rtpproxy to perfo
-- Original message
> From: Daniel-Constantin Mierla
> Date:05/05/2015 12:52 (GMT+05:30)
> To: "Kamailio (SER) - Users Mailing List"
> Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal
>
> Hello,
>
> do you have rtpproxy or rtpengine for r
(SER) - Users Mailing List"
Subject: Re: [SR-Users] Kamailio proxy for far-end nat traversal
Hello,
do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio is routing
only sip packets, you need the rtp relay application to help with media streams.
Cheers,
Daniel
On 01/0
Hello,
do you have rtpproxy or rtpengine for relaying rtp packets? Kamailio is
routing only sip packets, you need the rtp relay application to help
with media streams.
Cheers,
Daniel
On 01/05/15 13:15, rahul.ultimate wrote:
> Hello
>
> I need a small guidance on creating a light weight proxy whi
Hello
I need a small guidance on creating a light weight proxy which only forwards
the msgs to my sip server and also does supports symmetrical nated clients.
The way I have created the configuration is a slight modification of :
https://github.com/xlab1/sipfe_kamailio/blob/master/kamailio.cfg
BTW... Daniel,
you were 100% right... I had missed a record_route in part of my config
sigh..
just me being a complete noob with kamailio ..
thanks for your assistance.
Jay
On 12 March 2014 14:13, jay binks wrote:
> I have record_route and loose_route in my config ( record route , wher
I have record_route and loose_route in my config ( record route , where I
handle invites & loose_route within an if ( has_totag() ) )
Im not sure what im missing here, this dosnt seem to cause the contact
headers to be re-written
these contact headers do / can have the IP of the FS box ( behind t
Hello,
why you need to hide the ip in the contact if it is in sdp? You don't
really achieve topology hiding with it.
If you use record_route()/loose_route() in kamailio, the sip packets
related to active calls should come always to kamailio and kamailio will
forward them to freeswitch.
Che
So im using Kamailio as a load balancer and proxy in-front of a few
Freeswitch boxes.
Im not going to use RTP Proxy, as my FS boxes will be public on the
internet ( Direct RTP connection is fine ).
I want ALL SIP to go via Kamailio, but the issue im running into is that
the SIP Contact header has
In Asterisk - sip.conf
[MyKamailioUser]
type = friend
host = 192.168.2.2 - internal IP of Kamailio
insecure = port,invite
context = {where you would like to get calls - you can use default, or not
use this part at all}
In Asterisk - only authentication is based on IP of Kamailio server.
In Kamaili
Hi,
Kamailio is definitely the exact tool for this purpose, I have exactly the
same setup running as yours and for scalability we started using Kamailio
in front of our asterisk servers. Long story short, read these articles.
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-as
Hello,
We have been using Asterisk for sometime and over the last year have
started hosting instances for our clients on a vmware platform. These
virtual pbx are located on public ip addresses and each customer has
their own SIP trunk arrangements with various providers. We have
decide
On 14 July 2011 00:36, Iñaki Baz Castillo wrote:
> 2011/7/14 Sunny :
> > but not in this:
> > [UA] <-- [KAMAILIO] <-- [IP-PBX ]
> > because Kamailio is not keeping the location of the UA.
> >
> > I want to register UA's on Kamailio, but the authentication is done by
> > [IP-PBX]...
> > So Kamail
2011/7/14 Sunny :
> but not in this:
> [UA] <-- [KAMAILIO] <-- [IP-PBX ]
> because Kamailio is not keeping the location of the UA.
>
> I want to register UA's on Kamailio, but the authentication is done by
> [IP-PBX]...
> So Kamailio as Outbound Proxy + Location Server ... any idea on how I can
>
Hi Daniel,
I was using:
$ru = $(ru{re.subst,/[KAMAILIO]/[IP-PBX ]/});
t_relay_to_udp("[IP-PBX ]", "5060");
but your way looks better.
This works fine in this direction:
[UA] --> [KAMAILIO] --> [IP-PBX ]
but not in this:
[UA] <-- [KAMAILIO] <-- [IP-PBX ]
because Kamailio is not keeping the lo
Hello,
On 7/5/11 8:46 PM, Sunny wrote:
Hi list,
I'm trying to configure kamailio to work as a OBP (Outbound Proxy).
Basically KAMAILIO receive requests from UAs, after that it acts like
a client and send those requests to the IP-PBX.
On the same way the responses are forwarded from IP-PBX to
Hi list,
I'm trying to configure kamailio to work as a OBP (Outbound Proxy).
Basically KAMAILIO receive requests from UAs, after that it acts like a
client and send those requests to the IP-PBX.
On the same way the responses are forwarded from IP-PBX to the UAs.
[UA] --> [KAMAILIO] --> [IP-PBX
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