On 29/04/15 12:34, Grant Bagdasarian wrote:
That sounds possible! Thanks.
That does this description mean: “Execution time of load_gws()
function is O(N) * O(M), where N is number of different prefix lengths
and M is number of collisions for matching prefix(es) in lcr rules
hash table of
The Tag column size is 64. If I make this larger in the database, will it be
truncated once it is loaded into memory?
Through which list do I need to iterate? You mentioned the data is stored in a
hash table, what is the name of this hash table?
Basically for each call I need to call the
That sounds possible! Thanks.
That does this description mean: Execution time of load_gws() function is O(N)
* O(M), where N is number of different prefix lengths and M is number of
collisions for matching prefix(es) in lcr rules hash table of the LCR
instance.?
Does this mean it loads the
Hello,
On 28/04/15 20:42, Jon Bonilla (Manwe) wrote:
Hi all
I'm replacing an Asterisk based system with a kamailio based one. One of the
features the legacy system has is showing the subscriber the latency obtained
from the qualify option of sip.conf
Now, I'd like to measure the latency
On 29 Apr 2015, at 10:04, Daniel-Constantin Mierla mico...@gmail.com wrote:
Hello,
On 28/04/15 20:42, Jon Bonilla (Manwe) wrote:
Hi all
I'm replacing an Asterisk based system with a kamailio based one. One of the
features the legacy system has is showing the subscriber the latency
Hello,
if both phones are behind same nat and no nat processing enabled in
kamailio.cfg, the audio should work fine. Be sure there is no ALG in
your nat router or firewall. The best is to look at singaling, on the
server you can use:
ngrep -d any -qt -W byline sip port 5060
If the phones are
What about configuring two LCR instances with different lcr_id.
The first one can use only gateways with requested capabilities and the
second one all gateways.
Then you can make a decision about which instance to use during call
routing process providing this lcr_id to load_gws() function.
Am 29.04.2015 um 10:21 schrieb Olle E. Johansson:
On 29 Apr 2015, at 10:04, Daniel-Constantin Mierla mico...@gmail.com wrote:
it looks like you are the first wanting this, or at least the first that
has expressed it.
As Jon said, this is a feature that has been in Asterisk for a very long time
The Tag column size is 64. If I make this larger in the database, will
it be truncated once it is loaded into memory?
According to modules source code Tag's max size is hardcoded and will be
truncated. But this is not a bit problem. You can keep capabilities list
in htable and only put a
Hello,
On 29/04/15 10:06, Mihail Dakov wrote:
Hi all,
I am trying to run Kamailio server with IMS modules for which I have
gone through the following tutorials:
http://nil.uniza.sk/ngnims/kamailio-ims/preparing-debian-operating-system-kamailio-4x-platform
Hi
I've hit a problem with sht_rm_name_re() in htable module. I was calling it
like this:
sht_rm_name_re(Dlg=$var(callid)::tenant);
But when I used sipp to generate 600 concurrent calls for example, I called
this function when receiving BYE. But it removed more entries than it
should. Seems
Hi all.
I have this setup.
Trunk---KamailioFreeSWITCH
I have a trunk from a sip provided and registered successfully with the UAC
module. Incoming is working fine. I need to make out going through kamailio too.
I have it in the dialplan to forward the invite to kamailio from FreeSWITCH. I
Hello,
if i have two interfaces (eth0 and eth0:0)
I set kamailio to listen to the IP on eth0:0
This works great except when I try using TLS, kamailio routes traffic from
eth0:0 to eth0 then to correct destination
It should be just doing eth0:0 - correct destination
If I use UDP/TCP I am not
Hello All.
My English is bad so I hope you can understand.
I have been working with Kamailio some time, I following some of Guides of
http://kb.asipto.com, http://saevolgo.blogspot.com and http://nil.uniza.sk.
To Authenticate Asterisk sipusers I'm not have problems, but subscriber of
Mihail,
Are you able to share your configurations for each of the elements (P-CSCF,
S-CSCF and I-CSCF) ?
Cheers,
Abdul Hakeem
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Mihail Dakov
Sent: Wednesday, April 29, 2015 1:03 PM
To:
Hi Abdul,
I followed this tutorial: https://loadmultiplier.com/node/76 and the
configuration is the same but ips, ports and aliases. Have you tried that?
br,
m.dakov
On 04/29/2015 04:07 PM, Abdul Hakeem wrote:
Mihail,
Are you able to share your configurations for each of the elements
Hi Jibran,
Here is an old thread as reference:
http://lists.sip-router.org/pipermail/sr-users/2013-August/079336.html
I wouldn't want to do the whole handshake of INVITE,PROXY-AUTH,INVITE with
username/password on a Provider for huge number of calls..imagine sending
thousands of call to that
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