I think $fu does not content DISPLAY name. may be i am wrong...
2015-06-17 20:42 GMT+05:00 Al S :
>
> Hi,
>
> I am trying to change only the "display name" of the FROM header using
> uac_replace_from().
> However, the value is not being changed (see below)
>
> request_route {
>
>xlog
Hi Daniel,
Yes. I have *.htaccess* file in the root of siremis. I checked it in "
chathura@chathura-pc:/var/www/html/siremis$ " and confirmed it as,
-rw-r--r-- 1 www-data chathura 503 ජූනි 16 10:06 .htaccess
And of-cause I have run make prepare. I followed the steps in
*http://www.powerpbx.o
Daniel, see attached (attachment sent in a duplicate email sent just to
your miconda address earlier today 10:26 CDT -- I'm told you don't
routinely check that one). I added some logging with "RCK" in them. It
should be pretty clear what I'm logging with them. You can see r2 and
transport=tcp are s
On Wednesday 17 June 2015 15:31:33 Jonathan Hunter wrote:
> I am looking at a SIP redirect server, so I would scope on potentially 1 DB
> query per call to get routing information.
I have a couple for virtual redirect servers that do something like 200 cps
with multiple mysql queries (routing, to
Great thanks Guys that perfect.
Many thanks
Jon
From: bbrid...@o1.com
To: sr-users@lists.sip-router.org
Date: Wed, 17 Jun 2015 15:43:31 +
Subject: Re: [SR-Users] Performance Tests/Hard Recommendations kamailio 4.2
-200CPS
I currently have a 302 LCR redirect platform that we use for
I currently have a 302 LCR redirect platform that we use for dialer traffic
that utilizes a single MySQL server (with a redundant backup of course), does
17 queries of varying complexity for various routing data decisions (LRN
lookup, LERG data retrieval, routing table lookups, and so on), and h
Hi,
I am trying to change only the "display name" of the FROM header using
uac_replace_from().However, the value is not being changed (see below)
request_route { xlog ("L_INFO", "fu=$fu"); uac_replace_from("pbx",
""); xlog ("L_INFO", "fu=$fu");
# authentication route(AUTH); ..
On 06/17/2015 11:28 AM, Jonathan Hunter wrote:
No understood, I am aware kamailio is all about thousands of calls per
second, I just wondered if it had been bench marked, as many companies
unfamiliar with kamailio like these things :).
It's probably been a while since anyone's done some formal
Hi Juha,
I am looking at a SIP redirect server, so I would scope on potentially 1 DB
query per call to get routing information.
Thanks
Jon
> Date: Wed, 17 Jun 2015 18:22:56 +0300
> To: sr-users@lists.sip-router.org
> From: j...@tutpro.com
> Subject: [SR-Users] Performance Tests/Hard Recommendatio
Hi Alex,
Thanks for the response.
No understood, I am aware kamailio is all about thousands of calls per second,
I just wondered if it had been bench marked, as many companies unfamiliar with
kamailio like these things :).
I will look at it from the database angle as there will be the same amount
Jonathan Hunter writes:
> Has anyone recently performed tests on kamailio 4.2 running up to 200 CPS?
> It would be great to know the hardware requirements for this setup.
performance depends, e.g., on how many db queries your script uses when
it process one call. if no db queries are done, 200 c
In a purely hard-coded configuration, without external lookups or other
sources of I/O-wait, Kamailio has been shown to handle several tens of
thousands of CPS. By the time you hit 20k-30k CPS, you start to run into
hardware, scheduler and kernel-side limits that need tweaking.
The actually-ex
Hi Guys,
Has anyone recently performed tests on kamailio 4.2 running up to 200 CPS?
It would be great to know the hardware requirements for this setup.
Thanks
Jon ___
SIP Express Router (SER) and Kamailio (OpenSER) -
Hello,
It works, thank you.
Regards,
Seudin
On Wed, Jun 17, 2015 at 3:21 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> iirc, you can use sip_trace() in onsend_route(), there you can capture the
> outgoing ACK.
>
> Cheers,
> Daniel
>
>
> On 16/06/15 16:55, Seudin Kasumovic wrote:
>
> Hi,
>
Yes, i read the article about the developper's module.
There is a document relative to the SIP-I profil (with SPIROU):
http://www.fftelecoms.org/articles/la-fftelecoms-publie-la-v12-du-profil-d-interconnexion-sip-i
Le 17/06/2015 15:24, Daniel-Constantin Mierla a écrit :
Hello,
On 16/06/15 10
Hi,
A bulk registration has been added to the "location"
table:*** 1. row ***
id: 113 ruid: uloc-55808d5a-33bc-1 username: pbx domain: NULL
contact: sip:pbx@10.10.10.157:5060;bnc received: NULL path:
NU
On 13/06/15 15:26, Priyaranjan Nayak wrote:
> Hi All,
>
> I am using kamailio-4.2.5 for testing setup. After starting the
> kamailio I am getting below message
> /usr/local/sbin/kamailio -f /usr/local/etc/kamailio/kamailio.cfg -N 20
> -m 512 -M 8
> /usr/local/sbin/kamailio -f /usr/local/etc/kama
Hello,
On 16/06/15 02:00, Armen Babikyan wrote:
> Hello,
>
> I am running across the same issue as described here:
>
> http://lists.sip-router.org/pipermail/sr-users/2015-February/086796.html
>
> Was there a resolution to this issue? I come across this issue fairly
> regularly. Unlike the OP, h
Hello,
On 16/06/15 10:15, Jérôme Simionato wrote:
> Hello,
> I'm planning to interconnect kamailio with a SIP-I provider (an i have
> no more information at this time from my telco).
> I read the documentation about the SIP-I module, who need to be
> completed to match our needs.
>
> basically, we
Hello,
iirc, you can use sip_trace() in onsend_route(), there you can capture
the outgoing ACK.
Cheers,
Daniel
On 16/06/15 16:55, Seudin Kasumovic wrote:
> Hi,
>
> TM forward ACK statelessly (see log below), so there is no call of
> installed callback on TMCB_REQUEST_(IN)SENT in siptrace module,
Hello,
On 17/06/15 12:36, sharad tyagi wrote:
> Hi
>
> I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS.
> I have add and registered two IP phones with kamailio these phone able
> to call each other.
>
> How can i make outbound call with that I am using a asterisk server as
>
Hello,
do you have .htaccess file in the root folder of siremis? Have your run:
make prepare?
What operating system are you using?
Cheers,
Daniel
On 17/06/15 13:48, Chathura Senanayake wrote:
> Hello,
>
> Daniel, thank you for your response. I run it using web server and got
> the following lis
Hello,
Daniel, thank you for your response. I run it using web server and got the
following list. You can see that mod_rewrite is listed here..!
Array ( [0] => core [1] => mod_so [2] => mod_watchdog [3] => http_core [4]
=> mod_log_config [5] => mod_logio [6] => mod_version [7] => mod_unixd [8]
=>
Hello,
it sounds like mod_rewrite was not properly enabled by apache. Can you
create a php file with:
||
and run it through the web server and see if the mod rewrite is listed
there?
Cheers,
Daniel
On 17/06/15 11:18, Chathura Senanayake wrote:
> Hello,
>
> I need a help with siremis-4.2.0. I
Hello,
I need a help with siremis-4.2.0. When I try to login the the Siremis, it
shows a 404 Not found error message as follow.
Not Found
The requested URL /siremis/system/general_default was not found on this
server.
--
*Chathura Senanayake*
Associate Software Engine
Hi
I am novice to kamailio , I have installed kamailio-4.2.0 and SIREMIS. I
have add and registered two IP phones with kamailio these phone able to
call each other.
How can i make outbound call with that I am using a asterisk server as PSTN
Gateway. i am not famalier with kamailio script.
Rega
Daniel,
I'm reviving this old thread..
On 05/27/2015 01:40 PM, Andrew Pogrebennyk wrote:
> thanks for the answer, that's what I was thinking - maybe the flags do
> not persist or are destroyed after the per-branch failure route.
>
> However, the t_flush_flags description says this function can be
Hello,
I need a help with siremis-4.2.0. I have installed Kamailio-4.3 and have it
running. Then I installed Siremis-4.2.0. When I try to login to the Siremis
by using login name "admin" and password "admin", it shows a 404 Not found
error message as follow.
Not Found
The requested URL /siremis/s
t_relay() after loose_route() should simply use TCP if the second Route
has r2=on and transport TCP.
If not, send the log messages with debug=3 when handling the re-INVITE,
maybe you force send socket via some other functions.
Cheers,
Daniel
On 16/06/15 22:50, Ryan Kendrick wrote:
> After enabli
You have to upgrade to 4.0.7 as it replaced 4.0.6 in 4.0.x series. It
could be that the backtrace will still be relevant.
It is useful to know the parameters for registrar and usrloc modules.
Cheers,
Daniel
On 16/06/15 18:45, Marc Soda wrote:
> Unfortunately, the best I can do is this:
>
> Core
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