Hi Every One,
I registered in opensips voip services site. I found a tab regarding dialing
plan. such as *78 for enable dnd,*72 for setting to permanent redirect,*50
for voicemail inbox. I found no documentation in opensips regarding these
services. Is there a way for acheiving this thru opensips
Brad,
I use:
$var(rpidpriv) = $(hdr(Remote-Party-ID){s.select,1,>}{param.value,privacy});
if ($var(rpidpriv) == "full" || $var(rpidpriv) == "uri") {
setflag(10);# Privacy flag
}
Later in my script can check flag 10, and if it's set, I know it¹s a private
call.
- Jeff
On 11/24/09
Hi list,
Im trying to find the best way to detect RPID and then check to see if
privacy has been set. Ive checked the $re headers and they return:
sip:1234569...@x.x.x.x
Their is a function is_privacy() but your suppose to pass a privacy type
to it, from what I can tell their is no way to check i
I have put b2bua between OpenSIPS and Cisco, but still no luck. The
point here is that when fr_inv_timer hits, OpenSIPS prematurely sends
INVITE per the next branch and only after that CANCELs the previous one.
I don't think this is the correct behavior actually, and there was a
similar issue m
hi,
I am experementing with the presence.
when I use tcp, and i do unsubscribe all is well but with udp the server
doesn't acknolenge my unsubscribe.
any idea on how to test that? is there a way to see incomming msg on the
server?
thanks,
nir
___
Users
without the "listen" line (comment out)
listen = tls:192.168.2.100:5061
it's working
2009/11/24 Kristijan Vrban
>
> Just the default script. I only uncomment the tls settings, and added my
> internal ip
>
> /* uncomment the following lines to enable TLS support (default off) */
> disable_tls =
Just the default script. I only uncomment the tls settings, and added my
internal ip
/* uncomment the following lines to enable TLS support (default off) */
disable_tls = no
listen = tls:192.168.2.100:5061
tls_verify_server = 1
tls_verify_client = 1
tls_require_client_certificate = 0
tls_method =
Hello Anca,
it's done.
Thanks.
br,
takeshi
On Wed, Nov 25, 2009 at 1:59 AM, Anca Vamanu wrote:
> Hi Takeshi,
>
> Please open a bug report on sourceforge tracker
> http://sourceforge.net/tracker/?atid=1086410&group_id=232389&func=browse
> and specify what version you are using.
> Also can you pl
Hi Takeshi,
Please open a bug report on sourceforge tracker
http://sourceforge.net/tracker/?atid=1086410&group_id=232389&func=browse
and specify what version you are using.
Also can you please run 'bt full' to see also what are the values of the
variables there.
Regards,
--
Anca Vamanu
www.
I am configuring RADIUS with openser and CDRtool. when i dial a number it
loges into in mysql of RADIUS server but it gives the following error. and
when i check CDRs of CDRtool it shows all calls "inprogress".
Tue Nov 24 19:47:37 2009 : Error: rlm_sql (sql) in sql_accounting: stop
packet with z
Hello,
is the mailing list the place to report segfaults?
I'm doing some manual tests with presence and pua dialoginfo modules.
Opensips is crashing very frequently. Last time was this:
Log:
Nov 25 00:57:27 lab kernel: opensips[10924]: segfault at 0008
rip 2b2fe86c65f9 rsp 7ff
Hi Ross,
The dialog profiles are stored in DB only when a restart
occurs...Otherwise not.
Also, the profile info is loaded from DB only at startup and not during
runtime.more or less, there is not way to share the profile info
between multiple servers.
Regards,
Bogdan
Ross Beer wrote:
> H
Yes. Trying adding this to the reply route:
if (client_nat_test("1")) {
fix_contact();
}
For reasons I can't remember, the only test that works for the reply route
is "1", so you shouldn't change it.
Make sure you arm the reply route with t_on_reply() before t_re
On 15/11/09 22:38, Bogdan-Andrei Iancu wrote:
> Or maybe the INVITE from the biascica to yhe asymmetric is not going
> through because of the NAT (maybe the client is doing a poor job in
> keeping the pinhole opened in the nat). So, as time as the call is
> originated from behind the nat, it wil
Hello,
In Installation went fine but when i edit an entry in load balancer
module it gives an error "" Failed to issue query, error message: MDB2
error: syntax error
In the OpenSIPs.cfg file
xmlrpc module is loaded as
""
loadmodule "mi_xmlrpc.so"
modparam("mi_xmlrpc", "log_file", "/var/log/
Hi!
I´m using two Opensips as proxys, and they also take decisions about
redirections of different calls.
All messages pass through Asterisk.
In a call from 104 to 100, the first thing I do is send the INVITE message
to Asterisk. Later, after Opensips receives the new INVITE (from Asterisk),
it
Hi Bogdan,
I have solved the issue. I am using attended and blind transfers.
Using the dialog module I have added the calls to a profile using the
user name and then when another call comes in the script checks the
size of the profile for a given user.
There is a profile for each gateway, so whe
Hi Bogdan,
We experience the same thing under the same conditions (big amount of traffic
and rare errors):
Oct 27 06:18:05 ser10 /usr/sbin/opensips[2802]: ERROR:dialog:pre_match_parse:
failed to get From header
Oct 27 06:18:05 ser10 /usr/sbin/opensips[2802]: ERROR:uac:restore_from: new URI
sho
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