Thanks Daniel, works great!
Thanks again.
Jon
Date: Fri, 14 Dec 2012 20:31:58 +0100
From: mico...@gmail.com
To: sr-users@lists.sip-router.org
CC: hunter...@hotmail.com
Subject: Re: [SR-Users] Writing User-preferences Username into Radacct Table on
Call Forward
Hello,
Hi Daniel,
Thanks for the reply.
I am using that flag WITH_NAT and RTPProxy.
Please find my config file at http://pastebin.ca/2293600 and let me know I
have miss anything there.
Best Regards,
Roy.
On Fri, Dec 14, 2012 at 11:53 AM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:
> Hello,
Hi Daniel,
You do need a SIP stack to use this. We are using JsSIP here at Crocodile.
This has only been tested with Google Chrome so far, but there shouldn't be
anything browser specific in the stack.
Regards,
Peter
On 14 Dec 2012, at 19:59, Daniel-Constantin Mierla wrote:
> Hello,
>
> O
Hello,
On 12/13/12 11:07 PM, Peter Dunkley wrote:
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 usin
Hello,
communication with rtpproxy is done from modules/rtpproxy module -- look
inside that module sources.
Only RTP port is received to update the sdp, RTCP is just increasing by 1.
Cheers,
Daniel
On 12/14/12 1:11 PM, Austin Einter wrote:
Dear All
I am looking at Kamailio's module, file, f
Hello,
you have to do nat traversal for signaling and rtp relaying for media
streams. Default configuration file for kamailio includes the solution
for this case, using nathelper and rtpproxy. Looking into kamailio.cfg
for WITH_NAT token.
Cheers,
Daniel
On 12/14/12 7:37 PM, Raj Roy Ghandhi
Hello,
storing a new value in Radius acc record, should require to:
- define a new radius AVP name in the dictionary
- specify it in radius_extra parameter of acc_radius module, assigning
to it a name of an config avp
- in config file set that avp to the value you want
Here is an old tutorial
Hi,
I am trying to communicate 2 Jitsi clients in 2 separate private networks.
Both clients are behind NAT.
Kamailio Server is on Public IP.
Both Jitsi clients does register and presence works fine. And also text
messages works well.
But unable to call each other.
Had some one had the same issue b
Hi Roy,
Please keep emails on list.
Regards,
Peter
On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote:
> Dear Peter,
>
> Thanks for all the support that you gave me during the Kamalio
> configuration with RTPProxy.
> Currently it works fine with IP Phones but not soft clients. (Jitsi)
>
I did do some more tests, when a call is made didkamailio does not search
for the user in the location table, instead it searchs for user@domain, and
since the user is not on the same domain/ip it cant find the user. Se below
:
(31526) DEBUG: usrloc [udomain.c:575]: aor support1@didkamailio not f
Dear All
I am looking at Kamailio's module, file, function that receives rtp and
rtcp port from rtp proxy.
Can somebody please let me know which file/function I need to check.
Thanks
Austin
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users
Hi All,
I was wondering if anyone would be kind enough to let me know if they have
implemented anything like as follows in Kamailio?
We are looking to implement call fowarding, activating it as usual in
usr_preferences table.
I was wondering if on the outbound leg its possible to write the usern
Hello,
We are considering upgrading our VoIP platform and I'm wondering what kind of
hardware we need for our requirements.
The architecture will consist of a SIP Proxy Server (Kamailio) in front and a
cluster of Media/Application Servers behind it.
The requirements are based on a single server
Hi.Thanks for you reply!Provider is using Login and Password combination for
making calls.
--- On Thu, 12/13/12, Daniel-Constantin Mierla wrote:
From: Daniel-Constantin Mierla
Subject: Re: [SR-Users] From sip phone to provider trunk
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (
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