I just found something odd in the Debian package for kamailio-als-modules
and I'm looking for someone to verify it for me.
I verified that I'm using Kamailio 4.2
# Kamailio -v
version: kamailio 4.2.0 (x86_64/linux) 36e695
the docs
http://kamailio.org/docs/modules/4.2.x/modules/tls.html#tls.p.tls
I am trying to setup a really simple (I hope so) IMS platform, with
some basic services (like voicemail, IVR, videoconference), but I'm kind
of stuck.
I have installed Kamailio IMS on debian and a OpenIMS VM also. But my
little problem begins when I try to configure a application server (in
thi
Hi fellow kamailio users,
I'm playing actually with t_on_branch_failure, in a simultaneous forking
scenario ( lookup("location") returns multiple contacts ).
Any idea on how to get those "on_branch_failure" working ?
Thanks in Advance !
* Test by putting failure_exec_mode:
- at 1 => no branch f
On 05/19/2015 04:08 PM, Daniel-Constantin Mierla wrote:
> Debian/Ubuntu packages will be available soon as well.
Debian/Ubuntu packages available [0]
Official Debian package upgraded to 4.2.5-1 [1] ( will take some time to
refresh the info )
[0] http://deb.kamailio.org/
[1] https://tracker.debia
Greetings.
I want to modify INVITE answers processing in this way:
1) receive INVITE answer
2) check if recipient is registered but has no TCP/TLS connection now
3) suspend answer for 30 seconds (for example)
4) resume answer if REGISTER comes from lost client in these 30 seconds
5) terminate INV
Hello Daniel,
Thanks for replying back !
And please accept my apologies for responding late. I tried modifying the
configuration file to do a far-end NAT traversal but this case (wifi to 3G)
is still not working.
Below is the trace of what is happening on my systems -
Registration of UAC Behind
On 18/05/15 16:37, Klaus Darilion wrote:
>
> On 18.05.2015 15:41, Daniel-Constantin Mierla wrote:
>> Hello,
>>
>> On 18/05/15 14:57, Klaus Darilion wrote:
>>> Hi!
>>>
>>> Should log_prefix also work for xlog in reply routes? In my case it
>>> doesn't (Kamailio 4.2.2)
>> it should work for receive
On 18/05/15 03:53 AM, Sebastian Damm wrote:
Hi Alex,
On Thu, May 14, 2015 at 5:47 AM, Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:
According to the rtpengine module documentation for
rtpproxy_manage(), that's exactly what rtpproxy_manage() does:
http://kamailio.org/doc
Hello,
Kamailio SIP Server v4.2.5 stable release is out.
This is a maintenance release of the latest stable branch, 4.2, that
includes fixes since release of v4.2.5. There is no change to database
schema or configuration language structure that you have to do on
installations of v4.2.x. Deploymen
Hello,
I added the following xlog before function
*xlog("L_INFO", "[SDPOPS] executing function
sdp_remove_codecs_by_id($avp(s:codecs_to_remove)) ID=$ci\n");*
and if i have set debug=2 I can see the xlog message, if I change it to 3 I
cannot see my message, even if I set xlog with L_DBG i cannot
Hello,
can you enable cfgtrace via debugger module or add an xlog just before
calling the function in configuration file and see if related message
appears in syslog?
Cheers,
Daniel
On 19/05/15 11:05, José Seabra wrote:
> Hello,
> Thank you for your reply
>
> I ran kamailio with debug=3 and log_
Why not log the SIP message itself to the same syslog?
On May 19, 2015 12:46:34 AM EDT, jay binks wrote:
>So im logging lots of information to Syslog to assist our helpdesk in
>diagnosing customer issues.
>
>Im seeing LOTS of user agents that re-use the sip Call-ID ( especially
>in
>registers ).
I need some instructions on how to setup dbtext properly
I generated dbtext files through "./kamdbctl create"
I then add user through "./kamctrl add user1 user1"
It seems to have added the user because when I do the show './kamctl show
user1' command it displays
[1, ' user1', '10.1.1.1', ' u
Hello,
Thank you for your reply
I ran kamailio with debug=3 and log_stderror=yes and the only thing that i
see related with function sdp_remove_codecs_by_id is:
0(4707) DEBUG: [route.c:907]: fix_actions(): fixing
sdp_remove_codecs_by_id()
if i set debug=3 and log_stderror=no then i look for
Hello,
indeed there is no guaranteed unique token per SIP packet.
If one wants to identify the transaction, need to look at Call-Id and
CSeq, plus Via header (the via is relevant when there is a forking proxy
upstream).
Regarding the checksum, the upd checksum is not available, but kamailio
can
Hello,
short note that I am going to release 4.2.5 later today. If there is
anything to backport to branch 4.2, write an email to sr-dev and wait
for response. Once the announcement of the release is out, then
backports can be done as usual.
Cheers,
Daniel
On 12/05/15 11:33, Daniel-Constantin Mi
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