Sure, you can do that. That's probably one of its most common applications.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
T
Dear *All*,
I am new to voip world and interested in knowing if I we can use *KAMAILIO +
* *SIREMIS *for wholesale voip termination? We buy direct routes and give
it to resellers..currently using asterisk.
appreciate your support!
keep inspiring.
--
Regards,
*Abid Naseem*
+44 (793) 741 5043
+1
Thanks Anca,
No the records are not in the location db. I did modify the location
table and included "domain(string)" I dont know if that is necessary or
not but logically to me it makes sense.
However the records are not written to the table. I know for certain
that the user is authenticate
On 01/26/2012 08:21 PM, Krishna Kurapati wrote:
> Is there a configuration option to let kamailio use Public IP when
> setting record-route in 200 OK?
Of course, you need record_route_preset() - see
http://kamailio.org/docs/modules/3.1.x/modules_k/rr.html#id2667590
Also I would expect you need to
Hi,
I am having trouble with setting Record Route when Kamailio is behind a
Load balancer.
The Load balancer acts as a static NAT where the public IP of the load
balancer is mapped to
private IP of the Kamailio.
This is creating issues with ACK for 200 OK since Kamailio is stuffing the
200 OK wi
Hello,
I think that force_active affects the state of the SUBSCRIBE dialogs.
So, a presentity does not have to explicitly authorise a watcher. This
is not the same as automatically PUBLISH'ing presence state. I don't
believe force_active will do what you want here as it has nothing to do
with th
Hi,
but I've understood correctly. I only need kamailio and no other
software if I use the force_active switch?
Thanks
Jonathan
Von: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] Im Auftrag von Peter
Dunkley
Gesendet: Don
> > best practices regarding that as well?
> >
> > Andreas
> >
> >
> > ___
> > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> > sr-users@lists.sip-router.org
> > http://lists.sip-rou
Hi,
I am afraid I am not familiar with the configuration of those clients
for presence.
What should happen is that the client should PUBLISH at around the same
time as you REGISTER. When you change your presence information
(online/offline/busy, or your status message, or your avatar) the client
Hi,
I also found a memory leak in pua module which is probably causing the
increase in memory that you observe.
Laura, if you could please update your code and check?
Regards,
Anca
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users maili
Hello Peter,
the presentity table is empty. I have not seen any publish request.
Neither using bravis or x-lite.
Is there something I can do to make the client send a publish request?
Or is my configuration missing something?
I got
REGISTER
401
REGISTER
200
SUBSCRIBE
407
SUBSCRIBE
202
NOTIF
Hi,
Looking roughly in the pua_xmpp module I found some error cases where
the memory allocated by libxml2 was not freed. I have fixed them now in
master.
Not sure though if there are related to the memory leak Laura observed.
Laura, can you check if there were any error messages in the logs
p
Hello,
The dispatcher:dst-down and dispatcher:dst-up events are run once each
time a destination goes up and down. So if you have three destinations
in your set, and they all go down, dispatcher:dst-down will run three
times.
You can use $rm to find out which SIP method was in the request that
c
Hello,
Are there any PUBLISH requests between your clients (particularly the
client for sip:2002@sip.intern.domain) and Kamailio?
Does the presentity table in Kamailio contain anything?
The active state NOTIFY should contain the presentity information stored
for the contact in the presentity tab
Hello,
Some of the memory leaks we found here at Crocodile were to do with the
use of libxml2.
Various calls to libxml2 functions resulted in libxml2 malloc'ing
memory. This means you have to call the appropriate xmlFree...()
functions in the Kamailio modules to free up this memory. In some
pla
Hello all,
I'm trying to set up kamailio as a sip server with
presence support for Bravis.
So far I got the call processing
running on either openser-1.3.2-3build1 or 3.2.0+lucid1 running.
Problem is the notification is not using Content-Type:
application/pidf+xml
My openser config is at http
On 01/26/2012 02:34 PM, laura testi wrote:
Hi Hennings,
thanks a lot for your answer.
I'm sending you other information about our Kamailio xmpp server.
in attach the "sercmd cfg.set_now_int core mem_dump_shm" output.
And here:
kamctl fifo get_statistics | grep mem
shmem:fragments = 13107
shm
Hi Hennings,
thanks a lot for your answer.
I'm sending you other information about our Kamailio xmpp server.
in attach the "sercmd cfg.set_now_int core mem_dump_shm" output.
And here:
kamctl fifo get_statistics | grep mem
shmem:fragments = 13107
shmem:free_size = 99724520
shmem:max_used_size =
http://kamailio.org/docs/modules/3.2.x/modules_k/rr.html#id2527009
On 26.01.2012 08:24, Uri Shacked wrote:
On Thu, Jan 26, 2012 at 9:24 AM, Uri Shacked mailto:uri.shac...@gmail.com>> wrote:
Hi,
i would like to detct the direction of the BYE. was it drom the
caller or the callee?
Hi,
My question concerns Call Control.
When a SIP call's timer ends, Call Control uses dlg_end_dlg MI command of
the dialog module to send a BYE to the UA.
Can I intercept the BYE message, drop it, change it and send instead an
INVITE or a REDIRECT using (maybe) the UAC module?
The INVITE/REDIREC
Hi David,
I believe it has to do more with your configuration fine than with
location and db_cassandra, but lets see exactly what happens.
First of all do a message capture on the server to see that happens to
the Invite the first time it passes the server. Look at the RURI. Is it
changed to t
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