Hello,
inaddr any is special as a single socket receiving all traffic, but then
it has its own limitations when attempting to send.
Kamailio is building the list of sockets at startup and use that at
runtime, so it is not possible to have a new socket added at runtime.
However, depending on what
Le Mon, 10 Aug 2015 18:33:47 +0200,
Daniel Tryba a écrit :
> Only strange thing is that $var(sdp) is set to 0 on on the second
> passing of below during branch routing after handling a 302 redirect
I don't know the full context here, but since $var()s are
process-local, is the process reading th
I'm unable to get rid of the \n in "multiline" content of a var.
xlog("var: $var(sdp)");
$var(sdp)=$(var(sdp){re.subst,/^a=rtcp:.*//});
xlog("var: $var(sdp)");
a=sendrecv#015#012a=rtcp:31543#015#012a=X-foo:bar
becomes:
a=sendrecv#015#012#012a=X-foo:bar
Using the s flag results in no substitution
Hi Alexandru,
Well the concept is simple: on INVITE, you will issue a special format
evapi request. You will receive the list of suppliers in the reply
(which is received in a separate route) and you continue processing from
there. These suppliers are simple tags so your creativity will be the
Thanks!
2015-08-11 13:19 GMT+03:00 DanB :
> Hi Alexandru,
>
> Well the concept is simple: on INVITE, you will issue a special format
> evapi request. You will receive the list of suppliers in the reply (which
> is received in a separate route) and you continue processing from there.
> These suppl
Hi Sandeep,
what is the problem here ? Kamailio just sends a 404 on its own or is
really sending calls to MSC and MSC is replying with 404 ?
On Mon, Aug 10, 2015 at 12:33 PM, Sandeep Chakravarthi <
ivschakravar...@gmail.com> wrote:
> Hi ,
> Sorry for the delayed reply.
> I have configured my Ast
Hi,
Kamailio is sending 404 Response and its not MSC.
If you see the pcap file , Kamailio has to forward the SIP invite packet to
MSC which it got from Asterisk server. But it is not happening.
I am attaching the pcap one more time for your reference.
In my pcap, below are the server details
172
Le Mon, 10 Aug 2015 16:25:36 +0300,
Juha Heinanen a écrit :
> i think it would be useful especially if sdpops functions could also
> be made operate on the pv. then there would no need to call
> msg_apply_changes(). the pv could be initialized from $rb and when
> all calls that manipulate the b
Thats because your configuration file is not sending packet out (RELAY) to
MSC instead it is only doing a Loadbalancer / destination lookup in
TOASTERISK route and comes out of it, processes the following routes in
order
route(SIPOUT);
route(PRESENCE);
route(REGISTRAR);
route(PSTN);
route
Yes, You are right and done the changes as you suggested.
Kamailio server is forwarding the call to MSC. But two issues are there.
1 .In the INVITE packet which is being sent from kamailio server to MSC, it
is coming Request-Line: INVITE sip:0730092190@*172.22.14.12*
That is my kamailio server
1 - Take a look at the Kamailio transformations and psuedo-variable page.
change the $td to the IP of the MSC; modify the $ru as $rU + "@
172.22.12.100:5060" where this is IP of MSC side.
2 - Wireshark guys could've said it SIP-3 - point is it doesnt matter at
this point since you know your MS
Hello,
i'm on my first try with kamailio. I need to build a SIP balancer that
should keep SIP
registration from VoIP provider and route the calls to the asterisk boxes
where an IVR
will take care to answer.
Here's my network topology:
+---> [asterisk1]
[publi
First of all I'd suggest to use
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
guide in combination with
http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html
But, assuming your platform is behind NAT, you need:
1st. Use rtpengine instead of rtpproxy
Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/
2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <568...@g
Thankyou Alexandru for your suggestions.
I'll give it a try tomorrow and will report my progress here.
It seems that i'm not so far from the result!
Bruno
Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi <568...@gmail.com>
ha scritto:
> Also, take a look at kamailio-advanced.cfg, there is
Camille Oudot writes:
> i've created a merge request on the master branch to add the rtpengine
> "input" SDP pv. I'm not sure if modifying sdpops to work also on
> variables would be straightforward, but anyway, most operations can be
> done using transformations.
thanks. yes, i looked at some s
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