Hello,
yes, skipping the nat processing actions for requests coming from
presence server is good solution.
Cheers,
Daniel
On 12/12/12 9:36 PM, Owen Lynch wrote:
Hi,
we have 2 separate instances of kamailio running as a SIP proxy and
presence server. The proxy uses the nathelper module to c
Hello,
On 12/12/12 4:19 AM, Jon Morby wrote:
Hi
I'm trying to integrate a (K) front end cluster with an Asterisk back end
cluster and Asterisk RT (legacy system)
I've followed the recipe at
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
(with one minor exception
Hello,
thank you. Maybe you can add it to the tracker or to the wiki, so it
does not get lost in archives and others can review/contribute.
At some point we probably have to add it to the git and use it for
packaging, but I am not that experienced in the debian distros to say
that it is the
Hello,
is your trunk provider requiring a username/password for the calls sent
to it, or it is just IP based peering?
Cheers,
Daniel
On 12/10/12 4:52 PM, andre second wrote:
Hi,
I have some sip phones and using them to register at Kamailio which is
located behind 2 asterisk servers. There 2
Hello,
in the load balancer you have to do nat detection and use path module.
The rtp relaying can be engaged by the proxy.
Cheers,
Daniel
On 12/8/12 2:51 AM, Nguyen Anh Tuan wrote:
Hi everyone,
I have 1 Load Balancer and 2 SIP Servers, all of them are Kamailio.
Does anyone suggest how to
Hello,
presence rules are interpreted by presence_xml module.
Maybe this tutorial can get you started, it is a bit old, but still
something to begin with:
http://kb.asipto.com/kamailio:presence:k31-made-simple
Cheers,
Daniel
On 12/7/12 8:51 PM, Dmytro Bogovych wrote:
Greetings.
I try to ad
Hello,
can you dump the dialog attributes (via mi/kamctl or from database) for
the two cases? Just to see what is stored different there.
Cheers,
Daniel
On 12/6/12 3:55 PM, Pavel Miskov wrote:
Hi all,
I want to end dialog (Kamailio sends BYE to both parties) that is
lasting longer then som
On Thursday 13 December 2012 12:09:46 Daniel-Constantin Mierla wrote:
> > The problem I'm seeing currently is that when a call is passed down a SIP
> > trunk to an end user on the (K) platform we're losing the DNID
> >
> > Asterisk delivers the call to SIP/account/DNID
> >
> > (K) however just tr
On Tuesday 04 December 2012 19:09:18 Daniel-Constantin Mierla wrote:
> Quick look over the sources, I can't see a reason of not changing the
> headers. Can you set debug=3 and then give the log messages printed when
> the re-SUBSCRIBE is processed?
Attached is a fairly minimal .cfg (depends on mys
Hi,
Crocodile has just open-sourced our MSRP over WebSocket (see
http://tools.ietf.org/html/draft-pd-msrp-websocket) Javascript stack. The
project is hosted on Google Code: http://code.google.com/p/crocodile-msrp/
The stack is distributed using the MIT License and was developed and
tested along
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