I will look at upgrading in the lab first. I need to stabilize this software.
What I notice is the ser process stops responding to external register
requests. This is why everyone is expiring out because the sip process decides
to stop responding. The cpu is pegged at 100% but at times it comes
Looks Great Jeremy,
I will see if there is of any help for me and let u know.
Thanks for sharing
On Fri, Mar 18, 2011 at 3:38 AM, Jeremy McNamara wrote:
> On 3/17/11 6:23 PM, Alex Balashov wrote:
>
>> In some cases, it can be, depending on what you're doing with Asterisk to
>> begin with. Pur
Am 16.03.2011 21:09, schrieb Steven Wheeler:
> $rd=$dd;
> $rp=$dp;
> $du=$ru;
This one I do not understand. Also I do not see the code where you
change $dU?
Anyway, it seems that 2 branches are added: maybe one by lcr module
internally via next_gw and one manually? You can try to change $dU
I am attempting to put together code which will allow us to redirect a
call to a different tn if the route failed.
Example:
User dials 11235551234
Kamailio uses LCR to route to 192.168.0.100 (404 error)
Change $rU to 1123555 in failure_route
The only way I can get $rU to show up in the SIP
Have no chance at all to help with the patch.
Will the patch be added to release 3.1.1 release?
/Stefan
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent: den 14 mars 2011 15:39
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Hi,
I have a service that requires a user to authenticate, in order to bill him at
the end of the service. I also need real-
time check of the balance during a call, in order to signal when the amount is
too low to continue the call. Is this
possible with SER?
Thanx,
Miguel Tadeu
Hello Carsten.
Still no lucky. I just moved the whole rtpproxy/ directory into 3.1 branch
and recompiled and then changed my script to use
rtpproxy_offer()/rtpproxy_answer().
Look:
Mar 15 14:04:38 devel rtpproxy[29111]: INFO:handle_command: new session
e2361a59b4588d51, tag ed382a18;1 requested,
Did any one had any problem with this?
I have a test setup where I have the following directives for dns caching:
use_dns_cache=yes#enabling DNS caching
dns_cache_del_nonexp=yes # deletes entries if the memory is
becoming full
dns_cache_flags=0
dns_cache_gc_interval
Hola
en la actualidad estoy montando un servidor kamailio que trabaja con un
servidor asterisk.
Entiendo que lo mejor seria que tanto kamailio como asterisk trabajaran
con los mismos usuarios, por eso el servidor kamailio autentica los
usuarios contra un servidor radius.
Este servidor r
Thank you Henning.
I posted to the wrong list, but thank you for your reply.
Regards
Deon
On Mar 18, 2011, at 3:40 PM, Henning Westerholt wrote:
> On Thursday 17 March 2011, Deon Vermeulen wrote:
>> Can the SPCE platform scale beyond 50 000 subscribers and 2000 concurrent
>> calls when adding
On Thursday 17 March 2011, Deon Vermeulen wrote:
> Can the SPCE platform scale beyond 50 000 subscribers and 2000 concurrent
> calls when adding more hardware, i.e CPU and RAM?
Hi Deon,
i don't know the specific setup of this plattform, but at least for kamailio i
don't think there should be any
Hi,
more than 3 millions calls have been processed and no problem (crash,
increment in memory allocation...) has been noticed since the update, so
this check works for us.
Thanks a lot,
regards
2011/3/4 Daniel-Constantin Mierla
> Hello,
>
> just committed a safety check for this case. If anyon
So, I've made resetup of network to run tests.
I've figured out, that one of the routers were changing SIP headers,
i.e. Contacts header to public IP. This caused UAC under this router not
to use STUN and gave one way audio.
Now I'm using fix_nated_contact and fix_nated_register. Location table
c
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