Crash is caused by calling abort() within the code. Apparently your scenario
ended up at the place where it was not expected. More detailed explanation
would be worth to read from one of the developers. Anyway, for the future you
can try using "mem_safe=1" in your kamailio.cfg ...
__
Hello Daniel,
Thank you for answer,
Regard my last message where Alex is answer me.
Can you please verify that this ldap authentication routing section is should
work. Because call between two registered extension not working at all I don't
see any attempts of negotiations, always get 404. I
Hello,
can you try the patch from the next commit?
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=2376c833aad92bf8661f985d5917d952874a7509
If all goes fine with testing, I will backport. The issue seems to
affect all/older versions, not only 4.1, but it happens in very
Btw)
24.03.2014 21:51 пользователь "Rainer Piper"
написал:
> typing error
>
> Looking for solution for external Asterisk subsribtion of presence states.
>
> should be ...
>
> Looking for solution for external Kamailio subsribtion of presence states.
>
>
> ;-)
> Rainer
>
>
> Am 24.03.2014 20:28,
Hi, i have installed the new version ok kamailio 4.1.2.
I don't know why Kamailio crash writing this ob the log:
: [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already
freed pointer (0x7fc3ab755eb0), called from dialog: dlg_hash.c:
dlg_update_cseq(591), first free dialog: dlg_has
Hi, i have installed the new version ok kamailio 4.1.2.
I don't know why Kamailio crash writing this ob the log:
: [mem/q_malloc.c:468]: qm_free(): BUG: qm_free: freeing already
freed pointer (0x7fc3ab755eb0), called from dialog: dlg_hash.c:
dlg_update_cseq(591), first free dialog: dlg_has
typing error
Looking for solution for external Asterisk subsribtion of presence states.
should be ...
Looking for solution for external Kamailio subsribtion of presence states.
;-)
Rainer
Am 24.03.2014 20:28, schrieb Alexandr Usov:
It is work for qualify - thanks.
Looking for solution for
It is work for qualify - thanks.
Looking for solution for external Asterisk subsribtion of presence states.
Found the Pinan projec in the web, but it seems only Asterisk 1.4 supported.
I needs for Asterisk 11 or 12 version.
2014-03-24 17:37 GMT+02:00 Daniel Tryba :
> On Monday 24 March 2014 15
Hello Alex,
Here output from dig.
Yes, for current setup I want setup as border gateway with couple redirection
and failover to DR site. Currently I got ldap authentication working. trying
test call between remote extension (which failing). And next step will forward
to voicemail which parti
Hi Rizwan,
that is the right approach .
For adding an Asterisk as SBC you can use the section route[PSTN] at
kamailio.cfg
#!ifdef WITH_PSTN
# PSTN GW Routing
#
# - pstn.gw_ip: valid IP or hostname as string value, example:
# pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address"
#
# - by default
I would like to report a bug on the presence module
kamailio 4.0.5
setting custom table names via
modparam("presence", "presentity_table", "kam_presentity")
modparam("presence", "active_watchers_table", "kam_active_watchers")
modparam("presence", "watchers_table", "kam_watchers")
does work, i can
Hi there. I'm a noobie to Kamailio and Freeswitch... but I'm trying to follow
the article located here:
http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc
I've tried to add all the sections marked with WITH_FREESWITCH in the sample
config in the article into my own kamailio
Thanks! Will try.
Can you tell me how I can get from Asterisk host hint status of Kamailio
registered peers?
asterisk> core show hints
11@ext-status : SIP/11
State:IdleWatchers 0
12@ext-status : SIP/12
State:Idle
On Monday 24 March 2014 15:23:31 Alexandr Usov wrote:
> Peers (from Internet behind NAT) registered on Kamailio (local ip
> 192.168.182.1), calls from/to routed via Asterisk (192.168.182.24).
>
>
> Can't use qualify info:
>
> <--- SIP read from UDP:192.168.182.1:5060 --->
> SIP/2.0 484 Address I
Is my question not well phrased? Or is too general? Can anyone help with a
document or an older thread which could help me?
Thanks
On Mar 24, 2014 1:57 PM, "Rizwan Khan" wrote:
> I want the following setup:
>
> 1 Kamailio server to handle internal calls (A/V), IM and Presence.
> 1 Asterisk or an
El 24/03/14 04:17, Daniel-Constantin Mierla escribió:
Hello,
I found some cases when variables were not freed, but I cannot test as I am not
using unixodbc.
Can you cherry pick the patch:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=edc78dfb148c22f0d256485193bbdb0185
Peers (from Internet behind NAT) registered on Kamailio (local ip
192.168.182.1), calls from/to routed via Asterisk (192.168.182.24).
Can't use qualify info:
<->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '
1efe8f023a80bfb343495d5c4f20ea35@192.168.182.24:5060' Method: O
Hello Daniel,
How can I find a doc about private memory? I think that I should also consider
adding PKG MEMORY.
What do you think of that?
Regards,
Igor.
De : Igor Potjevlesch [mailto:igor.potjevle...@gmail.com]
Envoyé : lundi 24 mars 2014 10:31
À : mico...@gmail.com; 'Kamailio (S
Hello Daniel,
Thanks for your quick answer.
Here is the output (but I don’t think that it’s still relevant because I had to
restart Kamailio couple of minutes ago to avoid these errors and the calls
drop):
{
entry: 0
pid: 3286
rank: 0
used: 1463104
Hello,
can you get the output of:
kamcmd pkg.stats
?
Also, if you have big config, work with database queries in config file,
consider setting more private memory with -M paramter. -m is only of
share memory:
OPTIONS="-P /var/run/$prog.pid -m 256 -M 12"
Cheers,
Daniel
On 24/03/14 10:23,
Hello,
After couple of weeks without issue, my Kamailio 4.0 has begun to reject
calls this week-end.
The first error : ERROR: [msg_translator.c:2012]:
build_res_buf_from_sip_res(): ERROR: build_res_buf_from_sip_res: out of mem.
OR
ERROR: [msg_translator.c:2012]: build_res_buf_from_sip_res(
Hello,
I found some cases when variables were not freed, but I cannot test as I
am not using unixodbc.
Can you cherry pick the patch:
-
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=edc78dfb148c22f0d256485193bbdb0185b76d2f
and see if not all goes ok? If not other issu
I want the following setup:
1 Kamailio server to handle internal calls (A/V), IM and Presence.
1 Asterisk or any other way to communicate with an NGN where I will create
the SIP Trunk to route calls outside of the network.
Is this the right approach or there is a way to directly communicate with
Hello,
remove the double quotes in the IF expressions:
if ("$avp(s:domain)" =~ "$fd") {
Values in between double quotes are strings.
Cheers,
Daniel
On 21/03/14 21:41, Slava Bendersky wrote:
Hello Everyone,
I am trying compare domain part of uri with ldap query result, getting
some syntax w
Web server was running and nothing seemed wrong on the server at quick
look, however, as I didn't have time for more checks, I issued a restart
of web server application after the first report, which might have
brought everything back to normal.
Thanks for reporting,
Daniel
On 24/03/14 01:42,
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