Hi all,
Running ser 0.9.6.
I notice these numbers going up and never going back down:
I am a bit concerened with the red text. Is there any reason why that does not come down to 0. Do I need to restart the ser process once a month or something around that timing?
So far I have not heard of any issues with user call setup but I just want to remain proactive.
Serctl moni outputs the following:
Server: Sip EXpress router (0.9.6 (i386/linux))
Now: Mon Mar 8 10:05:18 2010
Up Since: Fri Jan 8 11:24:17 2010
Up time: 5092861 [sec]
Transaction Statistics
Current: 1 (176 waiting) Total: 5624219 (0 local)
Replied localy: 6049354
Completion status 6xx: 0, 5xx: 121, 4xx: 308504, 3xx: 0,2xx: 5316846
Stateless Server Statistics
200: 13145729 202: 0 2xx: 0
300: 0 301: 0 302: 0 3xx: 0
400: 42 401: 0 403: 0 404: 350766 407: 16536998 408: 0 483: 0 4xx: 2
500: 1 5xx: 0
6xx: 0
xxx: 0
failures: 72
Michael Sabatella
IPC Systems
Network Services Engineer
Office: 1-201-253-2280
Cell: 1-908-380-5321
@ IPCGlobalNetworks.com<http://www.ipcglobalnetworks.com/>
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Hi, by running "kamctl fifo profile_get_size togw" I can see the number of
dialog tagged as "togw".
Also if I run "kamctl fifo profile_get_size togw 999888777" I can see the
number of dialogs tagged as "togw" with value "999888777".
Is there any way to display all current values for dialogs with a specific
profile/tag (i.e. "togw")?
The idea is building a simple script to display the current number of calls
per user (having that the value applied to dialog profiles means the calling
user).
Thanks for any suggestion.
--
Iñaki Baz Castillo <ibc(a)aliax.net>
Hello,
http://www.kamailio.org has a new look! The former site was built with a
cms system that discontinued the development, providing no longer
updates for newer technologies and nicer design.
We hope the new design helps to locate project resources easier and
makes site navigation more straightforward via comprehensive list of
links in the right sidebar.
Credits go to Elio Rojano for the logo artwork and to developers of GPL
Wordpress theme Cordobo Green Park which was used as start for the new
design.
If you are skilled in web design and can build a more personalized look
on top of wordpress, please contact us.
Cheers,
Daniel
PS. Do not forget, if you are tomorrow in London, UK, consider attending
Present and Future of SIP Routing to get latest update about the project:
http://www.kamailio.org/w/present-and-future-of-sip-routing-2010-london/
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
Hello,
I have a weird issue with Dispatcher module. When I do a ds_list I
get some of my destinations returned with flag=P. It appears that
when they are in this state, they are not included in the list of
destinations to send traffic to.
I have tried restarting Kamailio, and these certain servers always
return to flag=P.
I have confirmed the servers are up and running fine. They are
communicating without issue at the SIP and network level.
Any idea what could cause this?
Thanks,
Help. Please remove me from this mail list. Thanks<br><br>在2010-02-27 19:00:01,users-request@lists.kamailio.org 写道:
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>Today's Topics:
>
> 1. remove parameter (Elgar Onel)
> 2. Exotic case (but real) in which RtpProxy doesn't work
> (I?aki Baz Castillo)
> 3. Re: Exotic case (but real) in which RtpProxy doesn't work
> (Daniel-Constantin Mierla)
> 4. Re: Exotic case (but real) in which RtpProxy doesn't work
> (I?aki Baz Castillo)
> 5. Re: Exotic case (but real) in which RtpProxy doesn't work
> (Daniel-Constantin Mierla)
> 6. Re: Exotic case (but real) in which RtpProxy doesn't work
> (I?aki Baz Castillo)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Fri, 26 Feb 2010 03:00:25 -0800 (PST)
>From: Elgar Onel <elgaronel(a)yahoo.com>
>Subject: [Kamailio-Users] remove parameter
>To: users(a)lists.kamailio.org
>Message-ID: <444271.31134.qm(a)web114207.mail.gq1.yahoo.com>
>Content-Type: text/plain; charset="us-ascii"
>
>Dear friends,
>
>how to remove parameters from INVITE sip address?
>
>tia, e.
>
>
>
>
>
Hi List,
I see two modules in particular that I could really do with running on 1.5.3
- I have started testing 3.0 but will not put into production just yet.
Call control and topoh
Would it be possible to backport these to 1.53 ? Would there be much to
it.?. Call Control in particular is something that is very useful for us.
We will eventually move to 3.0 of course as it is amazing product with lots
of really nice new features just been burned badly before by moving to
something so new.
TIA.
Stephen.
Hello,
yesterday at CeBIT I've run into Fraunhofer booth and a banner attracted
my attention, photo at:
http://www.asipto.com/gallery/v/cebit2010/P3030721.JPG.html?g2_imageViewsIn…
According to the middle right bubble, 2000 was the first year FhG
presented SER at CeBIT. Given that, means the project should celebrate
(at least) 10 years these days?!?
I was among FhG staffers at CeBIT 2002, presenting latest SER at that
time. According to CVS repository on BerliOS, files such as Makefile and
main.c are dated 2001. Officially, SER was GPL-ed and open sourced in
2002 (year when ser project was registered to berlios.de software forge).
Andrei, more insights from non-open-source era? I got into the project
in January 2002, would be interesting to trace the birthday...
Cheers,
Daniel
I'm looking for a way to distinquish between locally generated 408s
triggered by the timers. My problem is that I'm using gateway failover
in a failure route and on 408s it would retry phones multiple times if
they didn't answer (no voicemail etc) - so fr_inv_timer was triggered
after receiving a provisional reply. I need to failover in case a
gateway doesn't respond with provisional (it's set at 5 seconds
currently).
Will this work, is there a cleaner solution? I haven't had the
oppurtunity to test it so I may have some syntax errors.
onreply_route {
if (t_check_status("1[0-9][0-9]")) {
xlog("L_NOTICE", "KAM_HOST: ($ci) $pr $mi route[$rm][0]
$fu TO $ru -> Provisional Response --> $T_reply_code\n");
setflag(FLAG_PR_RECEIVED);
}
}
failure_route[ROUTE_DP_FAILURE] {
if (t_local_replied("last") && t_check_status("408") &&
isflagset(FLAG_PR_RECEIVED)) {
xlog("L_INFO", "KAM_HOST: ($ci) $pr $mi
route[$rm][0] $fu TO $ru -> Local 408 after provisional response, giving
up\n");
exit;
}
}
Thanks!
Bob
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Hello,
I am pleased to announce a dedicated event in London, Tuesday, March 9
afternoon, where to present the version 3.0 and latest developments in
Kamailio and SIP Router projects. The event includes as well
presentations from various players in the VoIP and SIP market, more
details at:
http://www.kamailio.org/mos/view/Present-and-Future-of-SIP-Routing-2010-Lon…
It is a free event, the number of seats is limited, room being available
courtesy of Sun Microsystems (address 55 King William St, London EC4),
therefore the first come first serve policy is applied. You can also
enroll for a presentation, just send a short description about the topic
in the registration email.
Hope to see many of you in London next week,
Daniel
--
Daniel-Constantin Mierla
Kamailio SIP Router Masterclass, Berlin, March 22-26, 2010
* http://www.asipto.com/index.php/sip-router-masterclass/
I use SPA3102 with a iptel.org VoIP a/c, whenever my Firewall external IP
changed (SPA3102's WAN port only connected to the internal LAN with a static
IP, router function not used) the iptel.org VoIP a/c will report
registration failed. I have to remove the proxy domain field and re-insert
it then reboot to fix the problem, just wonder if this is a problem on the
SER side or on the SPA side. I am sorry if this is not a correct
appropriate place to ask such a question.