Dont want to be mean
I've also had to learn, but this is just log/message reading..
What does it mean...
/etc/init.d/kamailio: line 131: log_daemon_msg: command not found
Well, in the file /etc/init.d/kamailio on line 131 a command is called that is
not known
So you have to check the kama
Djemel,
Even though your file /usr/premises/etc./kamailio/kamailio.cfg might look ok,
even if it is perfect...
It is at the wrong location.
The init script expects the config file to be at
/usr/local/etc/kamaiilio/kamailio.cfg
So you either copy your config file to that location or edit the
Hello Chirag,
Ad 1. When making a test call do an ngrep on the kamailio: ngrep -d any -qt -W
byline port 5060
That information can help you to do more troubleshooting.
Ad 2. Enable the debug in the kamailion config : #!define WITH_DEBUG
Then start kamailio again, you will see a lot of info
Hello,
You can't replace Asterisk with Kamailio, well. depends on what you do with
Asterisk.
But remember Kamailio is a sip proxy server, Asterisk is a B2BUA.
As soon as Asterisk does your voicemail, transcoding, etc. Kamailio is not
an option.
Just google Kamailio vs Asterisk, you
Daniel,
Thank you for the examples.
Sorry, for the direct email, was not my intention.
Tried, and I get my results.
Thanks again.
Rgds,
Gertjan
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: vrijdag 23 augustus 2013 17:31
To: Gertjan Wolzak; 'Kam
Hello Vijay,
What kind of answer do you expect?
Rupayan has attached an example Kamailio configuration, you just have to
edit it to comply with your situation.
Or do you have other questions?
Please then ask the questions, elaborate on what you have tried and what are
the problems you run into.
readme:
http://kamailio.org/docs/modules/stable/modules/dialog.html#idp3737432
Is the afferent profile listing any dialog?
Cheers,
Daniel
On 8/23/13 2:50 PM, Gertjan Wolzak wrote:
Hello Carlos,
I have been looking at your module, cant wait to work with it, especially
the prepaid part.
But
ustomer's calls");
}
exit;
}
[1] http://kamailio.org/docs/modules/devel/modules/cnxcc.html
[2]
https://github.com/caruizdiaz/cnxcc/blob/master/example/kamailio-cnxcc.cfg
Regards,
Carlos
On Fri, Aug 23, 2013 at 5:16 AM, Gertjan Wolzak w
Goodmorning All,
I use the following route to check for concurrent calls by the same user, if
a concurrent call is tried it is not allowed.
route[CONCURRENT]
{
xlog("SCRIPT: Conccurrent call check");
if(!get_profile_size("caller","$fu","$avp(nrcalls)"))
{
for Kamailio and Asterisk.
Now i am creating some cfg file and testing on it. I don't have much idea
about cfg but trying understand each line one by one.
Thanks everyone for the support and reply.
Regards,
Nishar Hamsa.
On Mon, Aug 19, 2013 at 10:12 AM, Gertjan Wolzak wrote:
Hel
Hello Nishar,
Take a look at the Asipto Knowledgebase, they have good tutorials.
http://kb.asipto.com/asterisk:index
Good luck.
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Nishar M.H
Sent: maandag 19 augustus 2013 6:40
To:
e for
both. Do you think this is a network firewall issue?
On Tue, Jul 30, 2013 at 2:31 PM, Gertjan Wolzak wrote:
Hello Jason,
Well, as you can see now the error is:
U 2013/07/29 18:24:03.328456 <http://198.23.160.81:5060> 198.23.160.81:5060
-> <http://49.144.184.97:419
z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport
.
Max-Forwards: 70.
To: mailto:sip%3A639195015475@198.23.160.81>
>;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9.
From: "639178864952"mailto:sip%3A639178864952@198.23.160.81> >;tag=1b3b463c.
Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWEx
Hello Jason,
What you should do is on the Kamailio server do an ngrep, that information
will make it possible for the people on the list to take a look and help
you.
By only telling your experiences it is hard to help you.
Just give the command: ngrep -qt -W byline port 5060 and then
;
exit;
}
xlog("SCRIPT: caller value for $fu is now $avp(nrcalls)");
}
Maybe someone else can be helped by it.
Thanks.
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Gertjan
neous calls limit reached");
exit;
}
set_dlg_profile("caller","$fu");
get_profile_size("caller", "$fu", "$var(SIZE)");
xlog("SCRIPT: profile size is now $var(SIZE)");
rt of
prebuilt packages? Can you send the output of "kamailio -V"?
Thanks,
Carsten
2013/7/10 Gertjan Wolzak :
>
>
> Hello All,
>
>
>
> I am trying to have all dialogs terminated after one hour.
>
>
>
> So I have set the following parameters
>
>
>
>
Sorry
Important... Kamailio version 3.3
Rgds,
Gertjan
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Gertjan Wolzak
Sent: woensdag 10 juli 2013 8:58
To: sr-users@lists.sip-router.org
Subject: [SR-Users] send_bye dialog
io -c
loading modules under
/usr/local/lib/kamailio/modules_k/:/usr/local/lib/kamailio/modules/
0(10335) ERROR: [modparam.c:151]: set_mod_param_regex: parameter
not found in module
0(10335) : [cfg.y:3532]: parse error in config file
/usr/local/etc/kamailio/kamailio.cfg, line 495, column 33: Can't
Hello,
Just looking at the capture file, it shows that the Kamailio sends a 200 ok
on your invite.
The call never gets relayed to the Asterisk box.
Maybe you can take
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
as a guide, get this working and then start
Hello Lucky,
Think you will find what you need here: http://kb.asipto.com/asterisk:index
Rgds.
On 06/15/2013 04:39 PM, Lucky Santiago wrote:
Hi,
I am planning to deploy a kamailio as a front-end of a currently in
production voip gateway which is running in asterisk (a2billing) and i
can't p
Never mind...
Problem on my end not between keyboard and chair.
Rgds,
Gertjan
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Gertjan Wolzak
Sent: donderdag 30 mei 2013 11:39
To: sr-users@lists.sip-router.org
Subject: [SR
0/kamailio# git checkout
HEADmaster origin/HEAD origin/master
root@sip:/usr/local/src/kamailio-4.0/kamailio# git checkout
Regards,
Gertjan Wolzak
___
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Hello,
I looked at the asterisk config, looks like you have not configured that.
You need to add a dialplan in the extensions config and trunks in the sip
confiig.
If you want to make the dialplan realtime as well, you should add the dialplan
to the db.
Success.
Gertjan
-Original messa
Hello,
The port is configured in the sip.conf file.
Gj
-Original Message-
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of zhengyw
Sent: dinsdag 7 mei 2013 11:27
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Σχετ: Problem
am wondering, can that be caused by the failure route, or should I
be looking in another direction?
Hope someone can give me a pointer.
Thanks.
Gertjan Wolzak
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sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Thanks guys for the info.
Will try again and keep on it.
Rgds,
Gertjan
On 6/28/12 08:55:27 AM Daniel-Constantin Mierla wrote;
Hello,
On 6/27/12 6:08 PM, Corey Edwards wrote:
> On 06/26/2012 08:36 AM, Gertjan Wolzak wrote:
>> Which timer do I need to use
Hello All,
Situation, sip client on a smart phone, with a minimum registration time of
10 minutes, connected through wifi.
Just after a reregister I walk out of range of the wifi connection.
Then someone tries to call me, lookup location gives back that I am online
and can be found at i
Cc: Gertjan Wolzak
Subject: Re: [SR-Users] Alias possible for presence...
Hello,
On 6/20/12 9:19 AM, Gertjan Wolzak wrote:
Well, the account name is not public, but I would like to have the
possibility to have a "private" account for registering and a
"public"account to p
@gmail.com]
Sent: dinsdag 19 juni 2012 10:27
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Cc: Gertjan Wolzak
Subject: Re: [SR-Users] Alias possible for presence...
Hello,
On 6/14/12 10:28 AM, Gertjan Wolzak wrote:
Hello All,
I have the following challe
on on that?
Regards,
Gertjan Wolzak
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http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
Sorry, my bad
Missing dependencies
Regards,
Gertjan Wolzak
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efine FLT_NATS 5
#!define FLB_NATB 6
#!define FLB_NATSIPPING 7
Does anyone has an idea why te values are not loaded into the variables.?
With regards,
Gertjan Wolzak
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