Hi,
I want to do web (HTML5 + WebRTC) Sip client which can do the
video conference with multiple users.
Current release of SIPML <http://www.sipml5.org/> does 1 to 1 call.
I have no idea of conference with many users.
Is it the client that we need to modify to accept call and join
the conference ?
Do I need to send INVITE with extra parameters ?
Please advice me.
Best Regards,
Roy.
Hi all,
I.m. a newbi in Kamailio.
My problem:
I.m. a blind user and i programmed a voicechat in Freeswitch.
3 have 2 sip providers.
vodafone.de sipgate.at and freenet.de.
The problem:
i must register these proviers in kamailio.
And forward all incoming calls to Freeswitch,
and outgoing calls from freeswitch to kamailio.
Dtmf modes are: auto rfc2833 and inband.
Then i call:
dorf(a)dorf.blindi.net
i like to forward these calls to my freeswitch box.
the kamailio ip ist:
217.172.180.108
and the freeswitchbox:
217.172.170.120
Can your help me please?
Thanks.
---------------
Du kannst mich jederzeit kostenlos per Festnetz erreichen unter:
http://www.blindi.net/callback
homepage: http://www.blindi.net
blinde-misc mailingliste für blinde. anmeldung unter:
http://www.blindi.net/mailman/listinfo/blinde-misc
Hi,
My Kamalio development version works very well with websocket and webrtc
clients.
But when I try to call the guy in remote area (he had connected to the same
server with 3G dongle) no voice and video.
Here is how I have set it up.
1. Kamailio 3.4 development version running on public IP
2. NAT Traversal is done with RTPProxy 1.2.
3. IP Phones work very well. (phones are behind NAT)
4. Web page with WebRTC works well in LAN behind the NAT
But I try to call a account which in logged into same Kamailio server we do
not hear voice nor media.
I have attached the sip capture into 2 files
1. LAN webrtc client->LAN client web page call
2. LAN webrtc client -> 3G Dongle webrtc client
Please help me out to figure this out.
Best Regards,
Roy.
Hello,
I was comparing modules_s/timer with modules_k/rtimer to spot the
differences.
modules_s/timer is working only with core timers (fast and slow timers),
while the modules_k/rtimer can start own timer processes, or register a
task to main core timer.
Other differences ...
modules_s/timer:
- run on milisecond basis
- option to enable/disable timer
modules_k/rtimer
- run either on second or micro-second basis
- option to execute the many route blocks on same timer
So, apart of enable/disable timer option, rtimer can do more than what
timer module offer (mili-seconds can be run as 1000x micro-second).
Enabling/disabling a timer can be workarounded from script, using a
shared variable tested at begininning of rtimer route, like:
route[RTIMER] {
if($shv(rtimer)==0)
return;
...
}
Enabling will mean setting $shv(rtimer)=1, either from other parts of
the config or via MI/RPC commands.
Now, both modules can be kept, no naming conflict, it is just a matter
of code maintenance if we should keep timer module. Enabling/disabling
timers can be added to rtimer in the future, but it is unlikely to
happen before next major release (unless someone else does it).
Is anyone here using modules_s/timer module aware of other differences
comparing with rtimer?
Looking for comments to keep or obsolete modules_s/timer module ...
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
Kamailio SIP Server v3.3.3 stable release is out.
This is a maintenance release of the latest stable branch, 3.3, that
includes fixes since release of v3.3.3. There is no change to database
schema or configuration language structure that you have to do on
installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous
v3.x.x versions are strongly recommended to be upgraded to v3.3.3.
For more details about version 3.3.3 (including links and hints to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
Kamailio SIP Server v3.3.3 stable release is out.
This is a maintenance release of the latest stable branch, 3.3, that
includes fixes since release of v3.3.3. There is no change to database
schema or configuration language structure that you have to do on
installations of v3.3.0, v3.3.1 or v3.3.2. Deployments running previous
v3.x.x versions are strongly recommended to be upgraded to v3.3.3.
For more details about version 3.3.3 (including links and hints to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2012/12/kamailio-v3-3-3-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hello,
I am considering releasing next minor version from branch 3.3 at the
beginning of next week, Monday or Tuesday (Dec 17 or 18).
For outstanding issues, please use the tracker or send to sr-dev mailing
list. Developers that pushed fixed in the master branch should backport
them in 3.3 before the next week starts.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Hi all,
I want to end dialog (Kamailio sends BYE to both parties) that is lasting
longer then some defined time. everything is working fine when using
old fix_nated_contact function, but when using new add_contact_alias
function kamailio sends BYE to private addresses.
1. "kamctl fifo dlg_list" output when using fix_nated_contact():
caller_contact:: sip:bob@PUBLIC_IP:50784
callee_contact:: sip:alice@PUBLIC_IP:64472
2. "kamctl fifo dlg_list" output when using add_contact_alias():
caller_contact:: sip:bob@10.2.5.206:10000
callee_contact:: sip:alice@10.2.5.205:20000
dlg_manag() doesn't store contact alias for caller and callee so Kamailio
is sending BYE to private addresses. Am I doing something wrong or I can't
use possibility of ending dialog with contact aliases?
Thank you
Pavel
Hi this is Murali krishnan i am final student of master degree in Dalhousie
University, I am Working on my project, i m need of a sip server which can
support H.264 codec video calls...I created an account in iptel sip account
when i used the account it doesn't support video calls. If i use SER will
this support video calls?
--
*Regards
Murali Krish*
