I have been trying to install SER with mysql on my system (CentOS 4 -
running on a virtual machine)
i have installed mysql 4.1.20-1 and ser 0.9.6.
Now when i execute ser ([root@SER ~]# /usr/local/sbin/ser start)
I get the following error...
0(2174) ERROR: load_module: could not open module
</usr/local/lib/ser/modules/mysql.so>: /usr/local/lib/ser/modules/mysql.so: undefined symbol: PL_memory_wrap
0(2174) parse error (26,13-14): failed to load module
0(2174) parse error (149,28-29): unknown command, missing loadmodule?
0(2174) parse error (157,37-38): unknown command, missing loadmodule?
0(2174) parse error (163,32-33): unknown command, missing loadmodule?
0(2174) parse error (178,28-29): unknown command, missing loadmodule?
ERROR: bad config file (5 errors)
Originally there was no mysql.so file in the modules directory of ser,
so i copied the file from the perl-DBD folder....
I have no idea how i can make ser recognize the mysql module...?
Thanks for the help!
regards,
Christoph Reichl (christoph(a)arin.net)
Hi Ron,
this looks super. Please upload the module on the tracker (just to have
it there). In the mean while I forward your email on the users mailing
list so people can show (or not) their interest for this new
functionality. If there are no arguments against it, the module may be
uploaded on the public cvs - of course you need to take care and
maintain the module for the future ;).
Regards,
Bogdan
Ron Winacott wrote:
>Hello all,
> I would like to contribute the following new module to the OpenSER project.
>If you find it usesfull, great, if not, sorry for the spam :-)
>
>The new module gives OpenSER basic SIP Session Timer support. See rfc4028 for
>more information on SST. This new module uses the new dialog module to track
>the creation, updating and termination of SIP dialogs. The dialog module
>supports timed out termination of the dialog but at this time the timeout
>value is hardcoded (avp accessable and modparam() settable)
>
>What the sst module does is uses the dialogs own callbacks to update the
>dialog timeout value based on the current expire: header value.
>
>There is also a script function called sstCheckMin(min_se_value) that can be
>used in a proxy configuration to reply with a 422 "Session Interval Too
>Small" to a INVITE with a small MIN_SE value.
>
>As I see new functionality requirements of the module I will add it. If you
>have a request for some new functionality in the SST module, please email me
>your ideas.
>
>The attached file is a gzipped tar ball of the sst module directory tree. (cd
>openser/modules tar -xzf sst.tgz to extract)
>
>Thanks for your time,
> ronw
>
>
>
>------------------------------------------------------------------------
>
>_______________________________________________
>Devel mailing list
>Devel(a)openser.org
>http://openser.org/cgi-bin/mailman/listinfo/devel
>
>
Hi all,
I have problem to built serweb into openser 1.1.0.
i use openser 1.1.0 and it work succesfully. i can add user by using openserctl add. For web interface i use serweb 2004-07. When i tried add new user, i have error message when i click for confirmation. the error is:
400 ul_add: flags_expected
I check into the database (mysql), new user was added into subscriber and pending table but the aliases table was not..
When i tried add user by openserctl add, in serweb i can login as that user..
Why it happen?
What should i do to run serweb succesfully?
Thanks for your help..
Aldi
---------------------------------
All-new Yahoo! Mail - Fire up a more powerful email and get things done faster.
Hello,
thanks to a contributor with php knowledge, a patch to make serweb 0.9.4
working with openser is available at:
http://voip-info.org/wiki/view/OpenSER+1.1.x+and+SerWEB+0.9.4
The patch is fresh and not tested too much, but should allow to start
openser and serweb together and opens the framework to improvements by
other members willing to use the two applications.
If you get stuck at a point, post it to devel(a)openser.org along with
description and error messages -- maybe someone will be able to get a
workaround.
Cheers,
Daniel
Subject: Openser and my voip provider
HELLO
In my lan I would like to have three SIP phone registered whit my voip provider, with ip public and
Local Number Portability. Is it possible with Openser to circumscribe the voip local traffic into my lan, and to manage the rest of voip traffic in transparent mode?
Can my voip provider save the logs of the my voip local traffic?
Sorry for my bad english.
Thanks Advanced
Marco
Hello all,
I haven't been using OpenSER long so forgive me if this is a silly question.
We are currently using OpenSER 1.0.1 (no TLS) running on CentOS 4.2
and 4.3Linux, if that helps.
Basically I am trying to find out a way of checking for minimum and
maximum digit length dialled by our users, in order to filter some of the
traffic through our OpenSER server. Dialling rules aside, is there a way?
Perhaps something involving checking the length of the Request URI in an
INVITE message and then possibly sending a "404 Not Found" or similar
message back? Is there a particular module I should be looking at too, or
some generic functions that can do this?
Your thoughts and comments very welcome. Thanks in advance.
Max.
Hi.
I would like to use openser as a sip proxy under my provider NAT.
Here is my situation:
Sip client (software or hardware)
|
|
|
|
V
Milkfish with openser that's also the router and firewall that does NAT
|
|
|
|
V
Nat of my internet provider with public ip (fastweb.it)
|
|
|
|
V
Sip provider (skypho.net) (it has also a sipproxy)
Can I make it working? How? Because I tried but even registering fails.
bye
kysucix
Hi,
i've just notices that i'm getting
"ERROR:nathelper:nh_timer: out of memory" each time it passes this
section:
if (!allow_trusted() && client_nat_test("3")) {
setflag(7);
force_rport();
fix_nated_contact();
};
There are around 1.6GB of Memory free on this box, so i don't think
it's a real OOM error ...
--
Mit freundlichen Gruessen / Kind regards
Dominik Bay
Cablesurf Technik
Hi Users,
I'm new to Asterisk programming , I'm in working the Voip Technologies by
using the OpenSER for my call routing process and Radius For AAA.
But in Asterisk i need it for only PBX and VoiceMail,
For Account I'm using the Openser + Radius .
Main My doubt is that,
For Call Routing my using the OpenSER. every thing is
fine and Good ..
But i need the Voicemail , it forword to the Asterisk
Server, that . Is the Openser takes the Accounting part or Asterisk it Take
place ....
Who did the Database know the User had a voicemail in
his voice mail Box... That Databases i need How ?
Please Help me ....
And Mainly Excuse me in English
--
Thanks and Regards
*Ravi Prakash Sunkara*
*M*:+91 9985077535
*O*:+91 40 23114549
*F*:+91 40 40208727 *ravi.sunkara(a)hyperion-tech.com*
*www.hyperion-tech.com*
Hi,
I would like a little direction on how to add the sip_trace module for 1.1.0
to my openser 1.0.1 without having to figure it out all on my own.
I was thinking I would have to add some #include, #ifndef, #defines, etc.
here and there to pertinent .c , .h files and then run make from the
sip_trace directory itself and then move the sip_trace.so module with the
other modules to incorporate the merge.
Any advice would be very much appreciated but I do not want to upgrade to
1.1.x at this time.
Tracy
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