Hello,
I have a question on how to configure the dialog module ( 1.2.x from
cvs yesterday ).
With my config, ( attached) I can make calls and have verified that
the acc module is working correctly.
My question is, when I enable the dialog module, I can see that it is
incrementing call count correctly, but when a bye is received, the
dialog:active_dialogs statistic is never decremented.
In the debug level 9 logs, ( also attached) I see this error after the
200OK is sent to the bye:
1(969) DBUG:dialog:unref_dlg: unref dlg 0xa7ce5a98 with 1 (delete=0)-> 1
Is this a case of one of the timers being set too short? by the way
using a variable call length from well under a second ( using sipp )
to 20 second call doesnt' seem to make a difference .
Thanks,
Andy
Been trying to a fresh install of OpenSER-1.2 / Radius and i keep getting
the following error
on authentication , find attached . Googling has send me in circles i'm
dizy.
--
TC
Hi, all.
I have a Solaris 8 server which happily builds a nightly CVS build I
retrieved from CVS on 16/12/06.
When I heard that 1.2.0 had been released, I downloaded this tarball:
http://www.openser.org/pub/openser/latest/src/openser-1.2.0-tls_src.tar.gz
However, I'm having difficulties compiling it.
To avoid a couple of symbol referencing errors, I added:
-DDBG_QM_MALLOC \
and disabled:
#-DF_MALLOC \
in the DEFS variable definition in Makefile.defs.
I reached this by trial and error and by comparing the differences in
the file from the working nightly build I can compile and the latest
release tarball.
However, I'm still getting:
Undefined first referenced
symbol in file
warn cfg.tab.o
ld: fatal: Symbol referencing errors. No output written to openser
collect2: ld returned 1 exit status
gmake: *** [openser] Error 1
Can anyone put me out of my misery? ... I'm not a C guru and am
struggling a bit.
Peter.
Hi,
First of all congratulations to all the contributors for releasing 1.2.0.
I was just wondering if regular expressions can be used inside case " ----" syntax.
Example:
switch($ruri.user) {
case "1[0-9]" :
route(10);
}
Any help would be appreciated.
w/regards,
Jayesh
__________________________________________________________
Yahoo! India Answers: Share what you know. Learn something new
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Sorry, forgot to put the list in copy :-((
--
Regards,
-vma
.
---------- Forwarded message ----------
From: vallimamod abdullah <vabdulla(a)gmail.com>
Date: Mar 14, 2007 11:05 AM
Subject: Re: [Users] NAT issue with devel version
To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
Hi Bogdan,
On 3/10/07, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote:
> Hi,
>
> could you detail a bit what is not working, as it is not clear from your
> email ?
Thank you for having taken time to look at my issue. I finally managed to
solve it: It was not related to Openser but to my firewall who had some issues
with NAT handling.
--
Regards,
-vma
.
Dear all:
Could tell me to have the environmental parameter can only get the username of from header and the domain of from header?
ex:
caller: john(a)iptel.org
callee: mary(a)iptel2.org
use $SIP_OUSER => mary
$SIP_RURI => iptel2.org
$xxx => john
$xxx => iptel.org
exec_msg("echo reguest from $XXX | mail -s 'request for you' xx(a)xx.xx.xx");
$xxx =>username of from header
Thank you
changyu
2007/03/14
Hello,
It's been a little while since I last posted, but this
will look familiar to those who remember:
The scenario is this:
Phone sends openser an invite, which tries to set things up with the
PSTN gateway, which is unresponsive(*), and so it sends back to the
Phone:
SIP/2.0 503 Service not available
and the Phone sends back an ACK.
What should I do with that ACK?
One possibility is to do nothing, another possibility is
to t_relay() it.
What suprised me was that when I do nothing then another
"SIP/2.0 503 Service not available"
message is automagically sent to the Phone.
So, that can't be the right thing to do.
Meanwhile, when I t_relay() it then things usually work.
But this doesn't seem like the right thing to do, since the
ACK is meant for the very box on which I'm doing the t_relay().
So, what's the story?
Thanks,
-mark
(*)don't worry about why the PSTN gateway is unresponsive.
Hellos
I whant to run RTPproxy separatly from my SIP router, so I tried to run
it like this:
[root@vox rtpproxy-0.3]# rtpproxy -s udp:xxx.xxx.xxx.xxx:22222
rtpproxy: explicit binding address has to be specified in UDP command mode
[root@vox rtpproxy-0.3]# rtpproxy -l xxx.xxx.xxx.xxx: -s
udp:xxx.xxx.xxx.xxx::22222
xxx.xxx.xxx.xxx: = my IP
But when I run openSER, this happens
9(15400) WARNING: rtpp_test: support for RTP proxy has been disabled
temporarily
11(15407) ERROR: send_rtpp_command: can't connect to RTP proxy
I already tried with
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
and with
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock
udp:xxx.xxx.xxx.xxx:22222")
But it always says: send_rtpp_command: can't connect to RTP proxy
Whats the problem?
Im using this configuration in sip.conf:
http://siprouter.onsip.org/doc/gettingstarted/ch08s02.html
Regards
Joao Pereira
Hello,
I have working ser with all the db modules and lcr. I want to send useralias
as caller id in place of user name.
Say I have a ser account with the name testaccount, and an alias
15556667777 to the user testaccount. So if some one calls 15556667777 and
the call will go to testaccount.
Now if testaccount make an call then in the caller id I see testaccount. How
can I send 15556667777 as caller ID when, testaccount make a PSTN call.
I guess I can also use uri table and use the caller id from uri_user in
place of username. But I dont know how can I do that. I tried this,
modparam ("uri", uri_user_column", "uri_user")
where the default is set for username. But That did not help.
Thank you,
-Jai
Hi!
With help of google I found
http://www.openser.org/docs/avp_db_query.html which I could not find
linked somewhere. IMO this would be a good idea.
regards
klaus
--
Klaus Darilion
nic.at
