look at: http://www.opensips.org/Resources/PresenceServer- Original Message -From: "manal rifki" Date: 20.05.2009 13:02To: users@lists.opensips.org, uctimsclient-us...@lists.berlios.de Hello, i am trying to install presence server in my openimscore plateforme. i try
Hello list,
Will opensips act transparent if I its installed between UA and
softswitch which both supports overlap dialing?
For the softswitch documentation:
In this scenario, as soon as the minimum amount of digits is received
according to the numbering plan (after having received the minimum
Dear sir,
The opensips script coded with radius accouting,
if the call is terminated with BYE and 200 OK, radius stop accouinging
will be sent out.
I try a call without 200 OK for BYE, I made an UA discoonected with LAN.
acc module seems fail in radius accouting, no radius packert send out.
How
If I understand you correctly, this behavior is to be expected with a sip
proxy. I'm not entirely sure, but mediaproxy or rtpproxy may resolve this
issue for you. ie: something needs to know about the media stream as well.
This has been discussed on the list. -Brett
On Wed, May 20, 2009 at 7:23
Hi,
The links you put here are useless...try to post the acc content directly.
Regards,
Bogdan
Цэвээндорж ЖиМэйл wrote:
Hello,
I have configured acc information in acc table of mysql database.
Edit
One solution is to take a look at mediaproxy from AG Projects:
http://mediaproxy.ag-projects.com
Regards,
Norm
kaiser wrote:
Dear sir,
The opensips script coded with radius accouting,
if the call is terminated with BYE and 200 OK, radius stop accouinging
will be sent out.
I try a call
Thanks, I have used mediaproxy at the same time, why dom't I get benefit
of it? (of course with dialog module)
kk
Norman Brandinger 提到:
One solution is to take a look at mediaproxy from AG Projects:
http://mediaproxy.ag-projects.com
Regards,
Norm
kaiser wrote:
Dear sir,
The opensips
On Wed, May 20, 2009 at 7:46 AM, Iñaki Baz Castillo i...@aliax.net wrote:
2009/5/20 James Lamanna jlama...@gmail.com:
Hi,
I want to use OpenSIPs as the registrar (and NAT handler) for an
Asterisk/Trixbox installation.
I've got things partially working, but I've totally made a mess of my
El Miércoles, 20 de Mayo de 2009, James Lamanna escribió:
I'm using the basic NAT example with a little rewriting (as shown below).
The problem I have is that the SDP address is not being rewritten, so
on the asterisk box I see audio traces like this:
(x.x.x.x is the phone NAT IP address,
El Miércoles, 20 de Mayo de 2009, Uwe Kastens escribió:
Hello list,
Will opensips act transparent if I its installed between UA and
softswitch which both supports overlap dialing?
For the softswitch documentation:
In this scenario, as soon as the minimum amount of digits is received
Hi All, testing a new OpenSips(1.6.0dev0-notls (i386/linux) and OpenXCAP
Server using xcapclient and contained examples.
pres-rules PUT looks good:
[r...@y examples]# xcapclient -i pres-rules.xml put
put http://x.y.com/xcap-root/pres-rules/users/sip:888...@x.y.com/index
200 OK
etag:
hi server monitor,
This error is becos of the python-mysqldb which you are using doesn't
support the kind of stream data you are trying to send. If the
python-mysqldb(which supports and clears the following errors) is upgraded
there is a reconnect error. i am yet to find the solution for
Thanks Aswini, is there any scenario in which openxcap will run?
Is there another project out there that will let you test presence docs and
resource lists?
I have managed to get everything to compile on Fedora core and very hesitant
to start experimenting with mysqldb.
Does anyone have any
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