Hi Team,
Working on Kamailio setup with IMS components(P-CSCF, I-CSCF, S-CSCF).
Assigned IP's:
hss/server-10.0.0.9
P-CSCF-10.0.0.10
I-CSCF-10.0.0.11
S-CSCF-10.0.0.12
Here tried to send an REGISTER request from Boghe IMS client, the register request should go from P-CSCF to I-CSCF, here P-CSCF - I-CSCF communication is not happening but I could able ping.
One more thing observed from logs here is REGISTER request don't have username (username is not there in request line),
Execution log: 0(1338) DEBUG: ims_usrloc_pcscf [usrloc.c:106]: get_alias_host_from_contact(): no alias param
Please find attachment of Zone file, SCSCF.xml, Diameter.xml, ICSCF.xml and tcpdump.
Any changes/suggestions would be very helpful.
Regards,
Sainath
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Hello,
I am using module websocket and it works very well.
However I would like to send BYE to the other end if event
[websocket:closed] happens.
>From the docs I can see websocket module itself doesn't provide for this.
I was considering doing something like this:
- use module htable to match $si:$sp to dialogs
- use event_route[dialog:start] to insert dialog info to my htable under
$si:$sp of Websocket side of the call
- use event_route[dialog:end] to remove dialog info from htable
- use event_route[websocket:closed] to iterate over entries in the htable
under key $si:$sp and call dlg_get() and dlg_bye().
Obs: in the above, there is a risk of losing some dialogs as insertion in
htable cannot be done atomically, but I am fine with it as it it not
expected to happen as WebSocket users would only infrequently generate
simultaneous calls.
However before going with this, I would like to ask for other possible
approaches.
Regards,
Takeshi
Hello,
I would like some advice about a situation i have encountered recently. I'm using Kamailio 4.3.4 as a "proxy" : I have a media server, and to reach this server all the requests have to pass through my Kamailio server. The Kamailio server does not have to handle authentification or registration, all of that stuff is done on the media server. To do that I'm using the following route :
route[REDIRECT] {
if($rU == "" ) {
$ru = ("sip:1.2.3.4:xxxx");
}
else {
$ru = ("sip:" + $rU + "@1.2.3.4:xxxx");
}
}
1.2.3.4 is my media server's address. I have 2 users : an "internal" one registered on the media server, and an "external" one which is a random caller. It works fine for EXT -> Kamailio -> Media server requests but when I try to hang up with my internal user, the "BYE" request uses the following path : Internal user -> Media server -> Kamailio -> Media server instead of : Internal user -> Media server -> Kamailio -> External user. Obviously it is because of the REDIRECT route which send back the "BYE" request to the media server, when it should sends that request to the external user.
So, how should I modify that route to do what I want ? Maybe should I create a second route ? Any answer will be appreciated :)
Best regards,
Alexandre
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Thank you.
Hi All,
reference to this link:
https://www.kamailio.org/wiki/devel/rtcweb_breaker#scenarios
I want to know if the module to communicate with RTCWeb Breaker is
available or it was just a proposal and no more under consideration.
I have webrtc clients registered to Kamailio but due to lack of
(scalable/efficient) transcoding capabilities they can not make video calls
to Video IP-Phones.
I tried using webrtc2sip from doubango telecom and it actually enabled me
to achieve the goal, the problem with that case is webrtc2sip is working
with sipml5 client and there is not a big list of WebRTC clients that work
with it.
If I can achieve the referred rtc_web_breaker architecture then I believe a
lot of webRTC clients will be able to integrate with my setup.
Thanks,
Regards,
Sammy
Hi Kamailio users and Devs.
I am interested to know if any of you are using SST module with DIALOG
module and sharing the timeout avp between the two - expecting the call to
be torn down if no in-dialog requests are processed. If you are using it,
is it working as you expect?
To me it seems that the timers in the dialog module will never be updated
by any in-dialog requests. I just wanted to check before I start looking at
fixing... perhaps I am missing something.
Cheers
Jason
Hello,
based on a previous discussion on mailing lists, I just renamed the
dialog_ng to ims_dialog in order to reflect better its purpose nowadays
(use together with the other ims modules) and avoid confusion that it is
something that includes the features of dialog module and some new stuff.
Functions and params are not renamed, being anyhow overlapping a lot
with dialog module, but they are not intended to be used at the same time.
Perhaps some examples still need to be updated (contributions are
welcome here if you identify such places). If any other issue is
discovered to the ims modules, write here to get it fixed.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Book: SIP Routing With Kamailio - http://www.asipto.comhttp://miconda.eu