Hi Daniel,
I need to access custom database tables (i.e. not default with kamailio)
from a lua script (extending kamailio.cfg).
Processing any incoming SIP request starts with a lua function and that in
turn call other functions - and each one needs database access for taking
routing decision.
Hello,
you can use htable module to cache data in memory and access it from
config file or lua script via pv.
Cheers,
Daniel
On 6/13/12 2:27 PM, Sharif Tanvir Rahman wrote:
Hi Daniel,
I need to access custom database tables (i.e. not default with
kamailio) from a lua script (extending
Hello,
On 6/12/12 11:49 AM, SamyGo wrote:
Yes, I saw the event routes in new version and that seems relevant.
Yes deleting entries from hash table is definitely logical but what
about hash entries for calls which are in-call for like 4~10 hours !!
will hashtable delete those entries !
Hello All,
I have the following challenge.
We are using kamailio 3.2 and have enabled presence. Which works fine.
But we would like to be able to work with aliases, so that the account name
and presentation name can be different.
Any Ideas. Or did I miss the documentation on that?
Hi All
With the upcoming release of version 3.3.0, we have updated our kamailio debian
repositories.
* Debian 5.0 lenny will no longer be supported. This means that no more
nightly builds for this distribution will be triggered. We'll provide builds
for stable releases of 3.0 ,3.1 and 3.2
So I am thinking of not thinking on auto-expire or create any code to
corelate dialog/active call data with my hash table. Instead Just focus on
latest version and then try this all.This new dialog keepalive function
seems better.linking between my hashtable entry with the dialog is not
required,
Hi Daniel
I've tried the patch on my 3.2.2 and it doesn't crash now!
Thanks!
Yufei
On 13/06/12 10:55, Daniel-Constantin Mierla wrote:
Hello,
can you try with the patch from commit:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=c737ff95bb2e742981d81088169baa60d4605b85
Hello,
can you try with latest master branch -- it should be fixed by:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=1d89d7bea854c2e2c646b5d13ba215795325b50f
Cheers,
Daniel
On 5/29/12 5:52 PM, Daniel-Constantin Mierla wrote:
Hello,
finally had a chance to look over it.
Hello,
On 6/14/12 11:36 AM, Yufei Tao wrote:
Hi Daniel
I've tried the patch on my 3.2.2 and it doesn't crash now!
thanks for feedback -- I backported the patch to 3.3 and 3.2 branches.
Cheers,
Daniel
Thanks!
Yufei
On 13/06/12 10:55, Daniel-Constantin Mierla wrote:
Hello,
can you try
I know that you can set the TOS for packets sent from Kamailio in the config
section.
I got the question if you can read the TOS in the IP packet of an incoming
response or reply in the config scripts?
And can you set it per message/transaction?
/O
HI
I have tried the kamailio 3.3,
user 101(10.15.20.131) send subscribe to (10.15.20.137),
kamailio return 200 OK, but with contact header as:
Contact: sip:10.15.20.137:5060;transport=udp.
is it correct?
Since before subscription expired, the client will send the re-subscribe
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.
Now I have to setup a SIP server/SIP PBX in our Lab for test, does the SIPX
support these codecs
Hi all, I have a project for the 3G related, AMR and AMR-WB support.
I'm using the client develop suite from the PortSIP(http://www.portsip.com),
as their said
support the AMR, AMR-WB with RFC4867.
Now I have to setup a SIP server/SIP PBX in our Lab for test, does the SR
support these codecs and
When compiling kamailio v3.3, I am getting this error from sipcapture
and siptrace. I am compiling on solaris 10, 64bit Sun server with gnu
tools.
CC (gcc) [M sipcapture.so] sipcapture.o
In file included from sipcapture.c:76:0:
sipcapture.h:48:4: error: unknown type name
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