Hello list,
Hope you all doing well!
I am trying to use the extra attributes (xavp_contact) of the usrloc module
to save some additional info about the user.
I am setting the value before the save() and doing a call to registered()
(not lookup()) before trying to access these extra attributes. This works
fine with db_mode=3 but does not work in case of db_mode=0.
Does anyone knows if it should also work with mode 0? I was expecting it to
work but a call to kamcmd ul.dump show nothing related.....
I don't want to use the DB for the location purposes because I prefer to
have multiple servers using the dmq_usrloc (that works like a charm!) and
it conflicts with DB persistence of the location table (when the DMQ
replication happens, all servers (sharing the same DB) try to save the same
user info in the location table)
Thank you,
Kind regards,
Patrick Wakano
Greetings,
I'm doing some monitoring of my Kamailio in order to quickly identify
system failures and overloads.
In order to do that i have scripts that read the output of "kamcmd
sl.stats" and identify the rising of "500_replies", "5xx_replies" and
"6xx_replies".
However, i have rate limiting on my Kamailio, which generates "503" replies
when limit is reached, which generates an unnecessary alarm.
Is there a way to get more specific stats about the repleis generated by
Kamailio? If that isn't possible, will i be safe only monitoring "500" and
"6xx" replies? In my experience, when the server has reached a processing
or memory limit, the reply is always 500, but i'm not really sure about
this.
Thank you for your help.
Best Regards,
Duarte Rocha
Good morning,
I want to realise a push notification proxy to signal incoming calls from asterisk to softphones. Is Kamailio the preferred solution for this? I looked also at Flexisip and Asterisk scripts and was told by a seasoned professional that Kamailio is the preferable solution as its scalable. True?
I would be glad if someone could share his experiences.
Also can one Kamailio instance be used as an asterisk load balancer and also push notification proxy ?
I have been a heavy openser user a long time ago. I used first openser as my core routing engine but then switched to yate. Please excuse my rather broad questions. From my experience with Openser the devil is in the detail. So any hints on where this setup may cause problems would be great.
Thanks a lot for your input.
Gerry
Hello,
I am considering to split the app_lua module in order to move the older
style API available under Lua 'sr' module in a dedicated module (like
app_lua_sr). Later this module can be obsoleted and removed, once we
know that everything there is covered by KEMI KSR Lua module.
In this way, we make it more clear that KSR is the preferred API to use
as well have the app_lua smaller, for those using only KSR and not
needing 'sr' API.
Therefore I am starting the discussion to see if anyone has something
against or other suggestions. I cc-ed sr-dev to make developers aware,
but I think the best place for discussion is sr-users, being related to
how Kamailio is used, therefore reply only to sr-users.
Cheers,
Daniel
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda -- www.linkedin.com/in/miconda
Hi,
I want to implement selective transcoding, e.g. avoiding Transcoding
between a HD codec (G722) and a non HD Codec (G711).
In case the caller does offer HD codecs, this is fine, e.g.
if(sdp_with_codecs_by_name("G722,OPUS")) {
rtpengine_offer("codec-transcode=OPUS codec-transcode=G722
codec-transcode=PCMA");
} else {
rtpengine_manage("codec-transcode=PCMA");
}
Now I have the issue, if the callee only sends me G711, I don't want to
offer G722 or Opus to the caller. However, rtpengine_answer() does not seem
to accept "codec-strip=G722 codec-strip=OPUS", e.g.:
onreply_route() {
if(!sdp_with_codecs_by_name("G722,OPUS")) {
rtpengine_ manage("codec-strip=OPUS codec-strip=G722");
}
}
As a result, the SDP always contains the transcoding options, e.g. G722 and
OPUS. I always end up in transcoding G722 to G711, which is meaningless in
that case.
Any ideas on how to solve that?
(above examples very simplified)
Thanks,
Carsten
--
Carsten Bock I Managing Director
ng-voice GmbH
Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com
Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
(0)40-52 47 593-99
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock, Dr. David Bachmann
Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
Hi,
quick question:
If understood it correctly in the past (and it worked for me quite well
this way), I can simply call rtpengine_manage() for subsequent requests
without adding any additional information (e.g. used interfaces, RTP/SRTP
translation or anything like that), as this information is stored in the
internal hash table of the RTPEngine module during call setup and initial
offer/answer.
However, this does not seem to apply to the transcoding options. When I
just call "rtpengine_manage()" for a Re-INVITE of a transcoded call (in my
case AMR-WB to G722), the transcoding options are no longer applied. The
forwarded offer does not get the transcoded codecs added (e.g. in my case
G722 is not added to the list of AMR/AMR-WB from the phone).
Did I misunderstand something or is it a bug?
My RTPEngine is still on 7.2, but I will double check with more recent
versions today or tomorrow. However, I also found no commit messages
relating to this.
Thanks,
Carsten
--
Carsten Bock I Managing Director
ng-voice GmbH
Millerntorplatz 1 I 20359 Hamburg I Germany
www.ng-voice.com
Mobile +49 (0)179-20 21 244 I Direct +49 (0)40-52 47 593-40 I Fax +49
(0)40-52 47 593-99
Sitz der Gesellschaft: Hamburg
Registergericht: Amtsgericht Hamburg, HRB 120189
Geschäftsführer: Carsten Bock, Dr. David Bachmann
Ust-ID: DE279344284
Hier finden Sie unsere handelsrechtlichen Pflichtangaben:
http://www.ng-voice.com/imprint/
Hi Everyone
When P-CSCF receives INVITE from client it sends INVITE to S-CSCF then
S-CSCF without any look up(checking the user is registered or not)sends
INVITE to I-CSCF then I-CSCF sends LIR and receives LIA and sends INVITE to
same S-CSCF (in case of one S-CSCF exists).My Question is here why without
any look up S-CSCF sends INVITE to I-CSCF??this procedure affects system
performance and increases call set up time
Hello,
Can someone help me to discard those messages from kamailio log files?
ERROR: <core> [core/parser/parse_fline.c:262]: parse_first_line():
parse_first_line: bad message (offset: 22)
ERROR: <core> [core/parser/msg_parser.c:681]: parse_msg(): ERROR:
parse_msg: message=<HTTP/1.1 101 Switching Protocols#015#012Sia:
SIP/2.0/TLS 212.2.161.34:60477#015#012Sec-WebSocket-Protocol:
sip#015#012Upgrade: websocket#015#012Connection:
upgrade#015#012Sec-WebSocket-Accept:
VeiXbJClWneIrIyYgM5v3XKYHLA=#015#012Server: WebRTC
GW#015#012Content-Length: 0#015#012#015#012>
As I understand this is just an HTTP response to the first request, and
shouldn't go through config file execution, but I don't understand why it
is getting through it.
Thank you
Hello! I have setup kamailio+rtpengine+asterisk.. And I have issue with
early media for WEBrtc endpoints... As I understood right, for this case I
need to use rel100(PRACK) for sending early media... But I didn't find any
example for solve this issue... Can you help me with it? It would be
greatly appreciated!
