Hi Friends,
I am trying to use the Kamalio 4 and SIPML5 and so for it is working fine.
But when I subscribe for presentity of another then I receive all the
entries of that user in the presentity table in mysql db.
It seem that it I receive the entries that are not expired. Is there a way
to get the latest status of the user or just avoid duplicate entries in
the presentity table.
Please advice.
Best Regards,
Roy.
Dear All,
I have build and installed kamailio 4.0.1 from source(tarball
release) in Cent Os 5.8 in i386 machine.I can start kamailio but when i
want to connect kamailio using SipMl5 client which uses sip over websocket
the client can't connect with the server.I am attaching my generated
kamailio.cfg file with this mail.Is there any additional configuration
needed for websocket support in kamailio 4.0.1?I have located the example
websocket.cfg file in kamailio source but totally confused what
modification to be done in the end of the file(e.g Main SIP request routing
logic blocks).
Note: I moified *modules.lst and include db_mysql auth auth_db sl nathelper
tm xhttp tls msrp websocket modules
*
(re-sent due to bad English)
Hi,
I have 3 registered test users, how can I configure Siremis to do the trunk
to freeswitch using LCR or Carrierroute rather than using the code below. I
am keen to be able to setup Inbound + Outbond trunks via Siremis. Do you
know if there is a manual for Siremis or a how to / step by step?
if($rU =~"^01") {
$ru = "sip:" + $rU + "@__FREESWITCHIP__";
route(RELAY);
exit;
}
Currently with the above code if a user phones one of the other extensions
it tries to route out to the PSTN network rather than the extension, is that
because I have put the above code in the wrong place in the config so it
never gets to the code to route to the extension? (routing to PSTN is fine)
Or do I need an if else statement wrap checking if local user, please can
you give me some idea of the code ...
Thanks
Tony
From: Daniel-Constantin Mierla [mailto:miconda@gmail.com]
Sent: 20 May 2013 16:19
To: tony.turner(a)nodemax.com; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio + Siremis Outbound route
Hello,
if you want to send all calls that arrive to kamailio having the prefix 01
to freeswitch:
if($rU =~"^01") {
$ru = "sip:" + $rU + "@__FREESWITCHIP__";
route(RELAY);
exit;
}
Be sure calls are authenticated at that point and, if needed, the call is
not actually coming from freeswitch.
Cheers,
Daniel
On 5/20/13 11:33 AM, Tony Turner wrote:
Hi
Version Kamailio v4.0 + Siremis installed on Debian Wheezy via apt-get
install
I want to use Kamailio as a proxy edge register to our network.
I have installed Kamailio and freeswitch.
I can register on Kamailio but I can't route a call from my sip client from
Kamailio to freeswitch and out to PSTN
Sip client ---- Kamailio-----freeswitch-------SS7 ISDN SIP Gateway ---
Carriers
If I register direct on Freeswitch I can route out to PSTN but I don't
understand Kamailio routing.
Can someone let me how I route say from SIP client registered on Kamailio to
prefix 01% which goes out to Freeswitch
Many Thanks
Tony
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
Hello,
if you want to send all calls that arrive to kamailio having the prefix
01 to freeswitch:
if($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(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, San Francisco, USA - June 24-27, 2013
* http://asipto.com/u/katu *
http://kamailio.org/docs/ser-getting-started/SER-GettingStarted.pdf
I'm looking for the authors of this document to find out what license it is published under. It would be nice to be able to use it and update it instead of starting a new "getting started" from scratch.
Please contact me if you get this message!
Sorry for cross-posting.
/O
On 4/22/13 5:30 PM, Juha Heinanen wrote:
> Andreas Granig writes:
>
>> Let me know if this is stupid and/or a complete overkill, but what about
>> introducing some kind of dummy mode, where you'd pipe a message into
>> kamailio via stdin, and get the resulting message out on stdout (e.g. in
>> ngrep style with ip information as first line, plus the content
>> following), plus a dump of internals (e.g. vars, avps) on stderr as
>> they're being assigned, and once the message is processed, kamailio
>> would just shut down again?
> in addition to the message, you would need to be able to tell which
> ip addr it is coming from and which ip/port in kamailio it is going to.
> also, on the output side, kamailio would need to tell which
> proto/ip/port it would use to send the message out.
Details about going out can be printed on onsend_route. In this route
block, one can execute drop and nothing is sent to the wire. It can be a
config started with a special define specified with -A parameter.
Enabling debugger module with cfg trace should give valuable information
about what has been executed from config.
From what Andreas suggesting, printing the value of variables as they
are assigned is missing, probably can be added by hooking in the
interpreter when doing the assignment operation.
Grouping all above under some global/command line parameter can be
useful to make it easier to do a dry run.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Dear list.
when using registrar module with path support does the following code work:
add_path();
save();
or maybe the path is added only for outgouing messages and the save() won't
be affected ?
Hello,
So I want to setup a Kamailio SIP Proxy( version 4 ) that will do NAT
signalling handling. With the help of nat_traversal and path modules.
Attached is my kamailio.cfg file.
My problem is I want to do keep alive on users who are behind NAT, so I
called nat_keepalive();. The script will work fine, but as soon as I
enable either add_contact_alias() or fix_contact(), the keep alive
doesn't activate.
I modified nat_traversal.c to identify the issue and found that it is
comparing the contact header from the request and the reply to find the
expiration. Trouble is, the request contact header was taken from before
I called fix_contact() and the reply header has the contact from after
the fix_contact() call. So my traces show that on the return trip the
callback is called, I bet the stack trace looks like this :
1. Some method in tm
2. __tm_reply_in() defined at nat_traversal.c:1358
3. get_register_expire called from nat_traversal.c:1377 and defined at
nat_traversal.c:878
4. STR_MATCH_STR called from nat_traversal.c:937 and defined at
nat_traversal.c:77
STR_MATCH_STR returns false because the headers are different.
I added
else
{
LM_ERR("failed to match because request
%s is not reply %s", contact->uri.s, r_contact->uri.s ) ;
}
to the code and see :
May 21 12:18:52 kamailio-dev-nat /usr/sbin/kamailio[30163]: ERROR:
nat_traversal [nat_traversal.c:946]: failed to match because request
sip:username@mytld#015#012#015#012 is not reply
sip:username@myip:59080>;expires=60;received="sip:64.18.189.196:5060"#015#012Server:
CenseredUA 5.2-rc1#015#012Content-Length: 0#015#012#015
So as you can see the original Contact header has the server's domain
name and the response has the IP address that set by fix_contact().
So I think that nat_keepalive() should not be comparing the Contact from
the original request but using the contact header from the modified request.
Any ideas what to do next ?
Thanks,
David