Hi folks,
I would like to show a project my colleage José Luis Millán and me are
working on. It's about SIP protocol running in a web browser. When
RTCweb (media capable web browsers) becomes a reality both
technologies together will allow real SIP endpoints coded in
JavaScript running in a web browser and interoperating with real SIP
networks. Bye bye Flash and custom protocol gateways.
http://sip-on-the-web.aliax.net/
Cheers.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
Hi,
I am trying to create concurrent calls limitations.
two questions:
1. should i use dialog module? or is there a better module?
2. i was trying to work with the dialog module, but didnt understand how and
where to configure the commands and which to use.... any examples or ideas?
BR,
Uri
Hello,
it is time to set the time to release v3.2.0. From my point of view,
testing went very well, considering also that core and main modules were
not touched much during this devel cycle, the most of the efforts were
on new modules and application service components such as presence
server, Lua ...
A convenient date for me will be Tuesday, October 18, 2011, allowing to
work a bit on packaging and docs. If there are other preferences, jump
in this discussion.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
Hi
I am using Kamailio 3.1.5 . I am using RTP proxy also.
I have used default kamailio.cfg.sample fiile , and just added line #!define
WITH_NAT.
I have another Main proxy. I wanted all my signalling and media packets
should just pass through machine where Kamailio and RTP proxy are running.
With this I found, call is established, all signalling and media packets are
passing through kamailio / rtp-proxy.
So far so good.
One way audio stream (from called party to calling party) quality is good.
The other audio stream (from calling party to called party is very bad.
Did anybody face this issue? Please help me to sort out this issue audio
quality issue.
Regards
Austin
Hi,
is there any setting which could allow me to set maximum memory per module ?
As I am testing the carrier route module, I've added for testing purposes
100 000 rules sofar. When I start the kamailio, it gives me :
Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
[../../ut.h:702]: could not allocate shared memory from available pool
Oct 10 14:46:46 kamrouter /usr/sbin/kamailio[2182]: ERROR: carrierroute
[cr_rule.c:155]: could not allocate shared memory from available pool
Thanks.
Hi
Just a quick question. How memory is allocated in Kamailio code.
Few places I saw pkg_malloc function.
Is it the only function used to allocate memory or there are similar
functions present in kamailio.
Thanks
Austin
Dear List,
I am trying to enable msilo on Kamailio but I'm getting an error
ct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]:
tm:load_tm: Module not initialized yet, make sure that all modules that need
tm module are loaded after tm in the configuration file
Oct 8 11:30:30 localhost kamailio[13856]: INFO: sl [sl.c:163]: could not
bind tm module - only stateless mode available
Oct 8 11:30:30 localhost kamailio[13856]: ERROR: tm [tm_load.c:48]:
tm:load_tm: Module not initialized yet, make sure that all modules that need
tm module are loaded after tm in the configuration file
Oct 8 11:30:30 localhost kamailio[13856]: ERROR: msilo [msilo.c:333]: can't
load TM API
Oct 8 11:30:30 localhost kamailio[13856]: ERROR: <core> [sr_module.c:875]:
init_mod(): Error while initializing module msilo
(/usr/local/lib64/kamailio/modules_k/msilo.so)
Any ideas how can I fixed it.
Thank you in Advance.
Regards,
Mark Anthony C. Delfin
Hi
I downloaded 3.1.5 Kamalio source, did make and install. Tried to run, and
tested few basic scenarios, everything working fine.
I have a main proxy, want to use kamailio as intermediate proxy and my
requirement is all RTP packets should pass through machine in which Kamalio
proxy is running. I beleive RTP Proxy I can run in same machine where
Kamailio proxy is running with proper configuration .
For this I am referring,
http://nil.uniza.sk/sip/nat-fw/configuring-nat-traversal-using-kamailio-31-…
page.
In one of the steps it says "So, if we are using the Rtpproxy server with
default configuration, we have to open /etc/default/rtpproxy file and
uncomment following line regarding of udp socket, that will be sued for
interconnection:".
I tried to search rtpproxy configuration file, could not locate.
While making and installing Kamalio proxy, the commands I used "make
prefix=/usr/local/ all and make prefix=/usr/local/ install".
Can somebody point me why rtpproxy config file is missing.
Also the link I am refering to is correct or wrong one.
Best Regards,
Austin
Hi,
my kamailio server is receiving from some customers 3 identical INVITEs when
call is initiated (separated by 200ms). Those 3 INVITEs are making a big
problem with call_control:
WARNING: call_control [call_control.c:1156]: dialog to trace controlled call
was not created. discarding callcontrol.
That is why, the prepaid limit is not working at all in this case. This way
the user can hack the prepaid protection of the account. Otherwise the
call_control is fuilly functional.
Anybody experienced the similar problem? If so, how to resolve it?
Thanks,
Mino Haluz
