Hello,
the details for organizing the 10 years SER conference in Berlin were
settled. After the initial announcement the interest was quite high, by
now we have a good number of attendees and speakers. We are still open
for new speakers, with the target of showing use cases of Kamailio or SER.
The event will be hosted by FhG Fokus institute, where SER project was
started. It will be a day full of presentations, followed by an evening
social networking event. We hope to be able to provide the evening event
for free to everyone attending, for that reason we look for sponsorships
- if you are interested to sponsor, contact us and we will provide more
details about available options.
Registration is now open for attendees. Note that the number of seats is
limited, don't wait for the last minute to register. More details,
including the registration form can be accessed on the event's web page:
* http://sip-router.org/10-years-ser/
Looking forward to meeting you in Berlin, enjoy a great summer till then,
Daniel
--
Daniel-Constantin Mierla -- http://www.asipto.com
Kamailio Advanced Training, Oct 10-13, Berlin: http://asipto.com/u/kathttp://linkedin.com/in/miconda -- http://twitter.com/miconda
I have installed Kamailio 3.1.0 and I am trying to test the pipelimit
module with postgresql. I ran the kamdbctl to create the tables but it
did not create the default table for pipelimit. Is there a script
someplace that I am missing or do I need to create my own script to
create the tables? I have search the directories and did not find a
script for the tables.
Thanks
Nathaniel L Keeling
Hi,
Is there any way to reload the content of the aliases table without
restarting Kamailio?
I have this in my cfg file
if (lookup("location")) {
xlog("L_WARN", "Callee is online - - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
route(TO_USER_SIPCHECKS);
route(RELAY);
} else if (lookup("aliases")) {
xlog("L_WARN", "Callee is in aliases - - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
route(TO_USER_SIPCHECKS);
route(RELAY);
} else {
xlog("L_WARN", "Callee is offline - - M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
t_newtran();
t_reply("486", "Busy here");
exit;
}
And changes in the aliases table does is ignored until I restart Kamailio.
--
Morten Isaksen
Hi all,
I'm getting this error when using custom parameters and I don't why.
when I use #!define WITH_ASTERISK the script throws the below error and
when I remove the #!define WITH_ASTERISK the script loads without
errors. I'm running kamailio-3.1.4-2.1.i386.rpm on Centos 5.6
Does anyone know why this might be happening?
parse error in config file //etc/kamailio/kamailio.cfg, line 123, column
16: syntax error
0(2907) : <core> [cfg.y:3412]: parse error in config
file //etc/kamailio/kamailio.cfg, line 123, column 16: unknown config
variable
0(2907) : <core> [cfg.y:3409]: parse error in config
file //etc/kamailio/kamailio.cfg, line 123, column 17-19:
ERROR: bad config file (3 errors)
kamailio.cfg::: lines 122-127
#!ifdef WITH_ASTERISK
asterisk.bindip = "127.0.0.1" desc "Asterisk IP Address"
asterisk.bindport = "5070" desc "Asterisk Port"
kamailio.bindip = "209.15.226.21" desc "Kamailio IP Address"
kamailio.bindport = "5060" desc "Kamailio Port"
#!endif
# This follows as described here:
http://by-miconda.blogspot.com/2010/04/kamailio-30x-dyk-07-custom-cfg.html
Thanks in advance
Hello,
My experience has been with Asterisk, but for what I need to do, I think Kamailio may be a better solution.
We currently log inbound caller ID information for voip calls (voip app posts call info to a URL). Our users have to fork calls to a custom SIPURI and we log the call information for them.
We would like to be able to accommodate users that can't fork calls, and have to register with a provider (PBX or VoIP provider). We see two ways to accomplish this:
1) Act as a b2bua, the user would register with our server and our server would register with their upstream VoIP service (PBX or provider).
or
2)Act as a proxy and modify the REGISTER request, such that incoming call signaling traffic goes through our proxy and we forward the signaling traffic on to the user's device.
Can Kamailio easily do either of these options?
Asterisk realtime doesn't support realtime changes of remote registration information, can Kamailio?
Would proxying in such a manner be supported my most voip devices and service providers?
--
Eric Chamberlain
Hi!
Seems like I am blind. There is a stored procedure which generates CDRs
from acc table entries - but how/when does this stored procedure gets
triggered?
Thanks
Klaus
On Fri, Jul 15, 2011 at 12:00 PM, <sr-users-request(a)lists.sip-router.org>wrote:
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> than "Re: Contents of sr-users digest..."
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> Today's Topics:
>
> 1. dbaliases for ring groups, possible? (Henrik Aagaard S?rensen)
> 2. Re: dbaliases for ring groups, possible? (Alex Balashov)
> 3. Re: dbaliases for ring groups, possible? (Morten Isaksen)
> 4. Re: xcap server - http load?balancer: xcap authentication
> problem (laura testi)
> 5. Re: bypass rtp traffic. (MingHon)
> 6. Re: AVPs missing in onreply_route (Sebastian Damm)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 15 Jul 2011 09:41:16 +0200
> From: Henrik Aagaard S?rensen <henrikaagaardsorensen(a)gmail.com>
> Subject: [SR-Users] dbaliases for ring groups, possible?
> To: sr-users(a)lists.sip-router.org
> Message-ID:
> <CAGH8SebPV=xz1n6Jp=M5fjnUhK0t=Za2aWd_jrpzbjdpdejatw(a)mail.gmail.com
> >
> Content-Type: text/plain; charset="iso-8859-1"
>
> I've read somewhere about using dbaliases for ring groups. So I can call
> one
> number and several different numbers will then be called. The first to pick
> up the call gets it.
>
> I've tried to an alias in my dbaliases table in MySQL, but I can only make
> it possible to call different numbers for the same user. In a way the other
> way around then what I thought.
>
> Is it somehow possible to call one number and then having Kamailio call
> several different numbers /users?
>
> As I'm very newbee to Kamailio it would be nice with a more detailed
> answer,
> then perhaps just "Yes, use the module ...".
>
Hello!
I noticed that the dispatcher's OPTIONs messages are not seen in onsend_route, even though I see responses in the onreply_route.
Any reason why I can't see these in onsend_route?
(Kamailio 3.1)
/O
