i am completely new to opensips but i am familiar with asterisk
here is my question
I want to Setup IVR application server
i want to handle maxium 1000 concurrent calls per sec
single call can of 1-2 hrs.
what configuration should i use for this.
I am planning to use asterisk as media server.
Dear All!
I need to replace Contact header receiving from UAC to
new one.
I have strange problem with all Yealink phones.
When I use
remove_hf/insert_hf in onreply route opensips didn't proper strip
header.
My config:
onreply_route[1]
{
if(is_present_hf(Contact))
{
Forgot to add, that problem exists on opensips rev 7915 and latest
opensips rev 8151 from trunk.
OS: CentOS 5.6 x86_64
Hope that's help.
On Mon, 11 Jul 2011 17:41:02 +0400, n...@uni-petrol.com wrote:
Dear All!
I need to replace Contact header receiving from UAC to new one.
I have strange
Hi,
quick guess, dont know why, but could it be related to insert_hf?
I always used append_hf which adds the header after the last header
field. I never tried insert_hf, append_hf worked fine for me.
BR
Max M.
Am 11.07.2011 15:47, schrieb n...@uni-petrol.com:
Forgot to add, that problem
Hi all,
OpenSIPS project will be attend to ClueCon on August 9-11, 2011 (
http://www.cluecon.com ), one of the most important VoIP events, for
both talking about OpenSIPS and VoIP anf for giving a short Quick Start
Training for OpenSIPS.
All details on both events can be found under:
Hi John,
The stats are kept in memory (and calculate based on the data cached by
opensips) so, the DB should not affect at all the stats.
What is the opensips version where this happens ? if the issue is issue
to reproduce, could you get full debug logs for all registrations ?
Regards,
On 07/08/2011 07:45 PM, José Pablo Méndez Soto wrote:
Thanks a lot Bogdan.
I follow the use of rPort all the way explained here. What I don't
understand, is, if the received and rport parameters are filled in by
the UAS that gets the request, how is that useful to the UAC that
originated it
Guys i want to check active call on my server how can i
i am using opensips.1.6.4-2
i tried dialog module but its not working
i am trying to export it to db sere it my config file
#
# $Id: opensips.cfg 7027 2010-07-15 13:48:29Z razvancrainea $
#
# OpenSIPS basic configuration script
# by Anca
try with dialog profile,
Dani
On Mon, Jul 11, 2011 at 10:53 PM, Akib Sayyed akibsay...@gmail.com wrote:
Guys i want to check active call on my server how can i
i am using opensips.1.6.4-2
i tried dialog module but its not working
i am trying to export it to db sere it my config file
#
#
For basic topology hiding the documentation says it is called by the
following..
if(is_method(INVITE) src_ip==10.10.10.10)
b2b_init_request(top hiding);
What I didn't see is where in the Routing Logic this is inserted and if
something else must be commented out, or is this simple added to
The latest development version of OpenSIPS has topology hiding in the
Dialog Module. That might be your best bet.
http://www.opensips.org/html/docs/modules/devel/dialog.html#id251279
On Jul 11, 2011 6:22pm, Mark Holloway m...@markholloway.com wrote:
For basic topology hiding the
Hi All,
I use a Panasonic NCP 500/1000 IP PBX for IP telephony. This PBX has two
interfaces, one for SIP (192.168.0.150:5060) and another for RTP
(192.168.0.151:12000+).
Currently, I'm using an OpenWRT box with siproxd working as a reverse proxy
(inbound = wan, outbound = lan) to allow
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