On Wednesday 07 October 2015 19:54:43 Daniel-Constantin Mierla wrote:
> Can you add log_prefix to print the callid, message type (request/reply)
> and cseq header with each log message? That should reveal what sip
> message is processed when such case happens.
Will do.
> I will look also at the
On Thursday 08 October 2015 10:03:52 Daniel Tryba wrote:
> Yes, the (shared) memory used shows the same trend as tm.current/number of
> dialogs. See attachment.
BTW the graph contains the kamcmd core.shmmem stats (total/free/used).
___
SIP Express
Hello community,
I work with Kamailio 4.2.0 which is placed between two networks (with two
interfaces) and RTPproxy in bridge mode.
The task from Kamailio is to handle the calls from internal networks to
internal networks only on the internal Interface without bridging. For external
Hi,
Other than this, I don't find any peculiar error message from Apache2
logs. Also the siremis/user/details/ doesn't load at all. Is
this a problem with the installation?
I would like to have a simple prepaid solution for my clients. ACC
works fine. Can anyone suggest a simpler solution to
Hi,
I set instant message store in kamailio using MSILO module. It works
fine in many cases.
We have special message handing (encryption in "session"), i do not go into
detail since it is not necessarily to solve the problem.
So there are situations, when the offline receiver should get a
Hello ;
Problem looks like blocked host and then gets connection error.
https://dev.mysql.com/doc/refman/5.0/en/blocked-host.html
how many user it has in subscriber table or did you check data of
subcribers. Maybe a wrong character makes connection errors so connection
lost.
--
View this
If the SDP is correct, then you might have specific issues related to your
specific deployment case. Snippets from others config files won't help. You
really need to investigate and understand your particular issue that you
are facing and fix it accordingly.
Regards,
Ovidiu Sas
On Oct 8, 2015
On 30-09-15 13:29, Fred Posner wrote:
Without a version of rtpproxy using the -A flag, you'll need to either
(1) update to a different version of rtpproxy or (2) skip rtpproxy and
have your asterisk handle all the rtp.
I tried rtpproxy v2, with the -A flag in bridge mode ( -A
Hi,
I found the culprit: it's a long Call-ID header. The header line is 113
characters. Why does it trigger this error even with a pv_buffer_size of
65536?
Is there some other buffer I can increase for large Call-Id headers?
-Sven
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Am 08.10.15 um 14:58 schrieb Sven Neuhaus:
> I found the culprit: it's a long Call-ID header. The header line is
> 113 characters. Why does it trigger this error even with a
> pv_buffer_size of 65536? Is there some other buffer I can increase
> for
Hello,
probably I will need to install siremis myself with user registration
enabled and see if works fine with latest php, etc., but I am traveling
at couple of real tme communication conferences, not having access to a
testbed for it.
What is your operating system, webserver and php version?
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