Hi,
I am thinking to write a module and export a function to be called from ser.cfg. I need the exported function to return a string value to be used by another function in ser.cfg. How do I do that in ser.cfg?
For example:
in ser.cfg:
function1(my_function());
in my_module:
str *my_function(){
...
return my_string;
}
Thanks in advance,
Ling
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Hi Yakout,
Check SerMyAdmin at http://sourceforge.net/projects/sermyadmin.
Regards,
Flavio E. Goncalves
----- Mensagem original ----
De: Yakout Esmat <yesmat(a)xtra.co.nz>
Para: users(a)lists.openser.org
Enviadas: Terça-feira, 1 de Abril de 2008 16:08:44
Assunto: [OpenSER-Users] My first OpenSER post
Hi All,
This is my first port on the OpenSER forum, I come from Asterisk world looking for scalability and redundancy.
I have managed to install openSER 1.3 for the first time yesterday on a CentOS 5 test server. I used SVN, which worked fine.
My questions is what is the recommendation in regards to web interface. I can see that we have 2 options SerWEB and Admin Web. If there are any links or leads on HOWTO install either one that would be of great help.
Which one is recommended for Admin and also for client interface?
Does any of them provide FULL admin functionality, as in creating extensions, configuring voice mail etc….?
Many thanks for your assistance.
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Hi,
I have a question regarding the way rtpproxy handles 'address filling'.
After a session has been created, the rtpproxy pre-fills the caller and
callee's addresses and we see that in the rtpproxy output like:
pre-filling caller's address with 192.116.246.234:41000
pre-filling callee's address with 192.116.246.234:20000
Then when it sees the actual packets coming in from a different source
port, it updates the address and we see it in the log like:
callee's address filled in: 192.116.246.234:1024 (RTP)
The audio then flows fine both ways.
My question is, what would happen it the actual packets came in from a
different IP while the rtpproxy was waiting between the state of
'pre-filling' and 'address filled' states? Will the rtpproxy accept
such a change that includes a new IP? Will it ignore packets from a
different IP and continue the session normally? Or will it abort the
session completely?
Thanks,
Andres
http://www.telesip.net
Hi Stefan
Thanks a lot for your prompt response. I can hear the announcement
now. Previously my audio setup was not correct.
Best Regards,
Frq
On Wed, Apr 2, 2008 at 12:11 PM, Stefan Sayer <stefan.sayer(a)iptego.com>
wrote:
> Hello,
>
> frq ser wrote:
>
> > Hi
> >
> > Thanks for your reply Stefan and my apologies for coming back to this
> > issue after such a long time. As you suggested, I included SDP headers in my
> > Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the
> > announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200
> > OK message, but no audio was being played. In the log, I got these messages
> > (just including the important messages)...
> >
> > ..
> > (17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK:
> > dispatch it! (17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying
> > '/usr/local/lib/sems/audio/192.168.3.208/200.wav <
> > http://192.168.3.208/200.wav>' (17358) DEBUG: getAnnounceFile
> > (Announcement.cpp:84): trying '/usr/local/lib/sems/audio/200.wav'
> > ..
> > ..
> > (17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global
> > address: 192.168.3.208 <http://192.168.3.208/> (17358) DEBUG:
> > negotiate (AmSession.cpp:262): remote party doesn't support telephone events
> >
> > Seeing the last line, I then used an Xlite softphone (instead of sipsak)
> > to contact SEMS. Again, the call gets established for about 10 seconds and
> > then just gets disconnected and no audio can be heard. Nothing in
> >
> To me the logs look normal. seems like it plays the announcement and then
> hangs up, as it should. the default_en.wav (or 200.wav, if you've copied it
> there), is only a few seconds long. are you sure that audio is setup
> correctly (volume etc), i.e. does the xlite test call work? if you call e.g.
> sip:conference@iptel.org <sip%3Aconference(a)iptel.org>, do you hear
> something?
>
> can you run tcpdump -i any or run wireshark to see whether packets flow
> between SEMS and xlite?
>
> you could also PM me a wireshark dump so i could have a look at what's
> going on.
>
> Stefan
>
>
Hi
Thanks for your reply Stefan and my apologies for coming back to this issue
after such a long time. As you suggested, I included SDP headers in my
Sipsak INVITE msg and the SEMS seems to like it. I am trying to play the
announcement using a INVITE message to SEMS at :5070. The SEMS sends a 200
OK message, but no audio was being played. In the log, I got these messages
(just including the important messages)...
..
(17358) DEBUG: execute (AmInterfaceHandler.cpp:172): Request OK: dispatch
it!
(17358) DEBUG: getAnnounceFile (Announcement.cpp:79): trying
'/usr/local/lib/sems/audio/192.168.3.208/200.wav'
(17358) DEBUG: getAnnounceFile (Announcement.cpp:84): trying
'/usr/local/lib/sems/audio/200.wav'
..
..
(17358) DEBUG: getCompatiblePayloads (AmSdp.cpp:348): using global address:
192.168.3.208
(17358) DEBUG: negotiate (AmSession.cpp:262): remote party doesn't support
telephone events
Seeing the last line, I then used an Xlite softphone (instead of sipsak) to
contact SEMS. Again, the call gets established for about 10 seconds and then
just gets disconnected and no audio can be heard. Nothing in the SEMS log
(see below) suggests that there is any problem (as far as I can understand).
I've checked that the file /usr/local/lib/sems/audio/200.wav is present and
I can play it directly in Linux using play command. (The file 200.wav is
just a copy of default_en.wav file)
I just want to know why the announcement file is not being played and why
the call between softphone and SEMS just gets disconnected after a while. I
am sure it would be something very basic :(
Thanks a lot for your help
Frq
Hello,
in the period of time April - June 2008, there will be a series of
openser social networking events organized in different places world
wide, during the time some other Open Source or VoIP events take place
in the same location. The events are most probably in the form of a
dinner, organized together with local representatives of openser
community. The events are intended to be rather small size, to allow
each one to know the others, in a nice and friendly discussion
environment. The primary goal is to strengthen the relations within
community and give opportunity to network for collaboration and
business. Participation to any of these events is free, everybody pays
for its expenses. Anyone interested in openser, sip or voip in general
is welcome to join.
Next events are:
- Bucharest, Romania, April 12, 2008, tied up with ROSDEV2008
(http://www.rosdev.ro - Romanian site only)
- Orlando, FL, USA, April 22, 2008, tied up with OPENSER-Asterisk SIP
Masterclass (http://www.edvina.net)
- Barcelona, Spain, May 6, 2008, tied up with OPENSER-Asterisk SIP
Masterclass (http://www.avanzada7.com/)
- Vienna, Austria, May 23, 2008, on the way to AsteriskTag
(http://www.asterisk-tag.org/wiki/Hauptseite)
- Berlin, Germany, May 28, 2008, tied up with LinuxTag
(http://www.linuxtag.org)
- Amsterdam, Netherlands, June 3, 2008, tied up with VoN.x Europe
(http://www.von.com/web/index.php)
The hour and place for each event will be announced few days before as
we know the number of attending people and we make the reservation.
Please send email to me to reserve a place, although you are welcome to
come at any time and without prior notice, you may find that there is no
more available seat to join the table.
If you live in the surroundings of these cities, and want to help with
organization, you know a nice and quite place suitable for the event,
let me know.
Cheers,
Daniel
--
http://www.asipto.com
So, I was kind of wondering how those using carrierroute where populating their routes.
I was hoping to build a cost table that contained the costs for all our routes and all our carriers, and then construct the carrierroute table as a view derived from the cost table. I guess my SQL (and those around me I asked about it) isn't that good, because I haven't been able to come up with a way to do it yet.
Doug.
____________________________________________________________________________________
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Hi All,
This is my first port on the OpenSER forum, I come from Asterisk world
looking for scalability and redundancy.
I have managed to install openSER 1.3 for the first time yesterday on a
CentOS 5 test server. I used SVN, which worked fine.
My questions is what is the recommendation in regards to web interface. I
can see that we have 2 options SerWEB and Admin Web. If there are any links
or leads on HOWTO install either one that would be of great help.
Which one is recommended for Admin and also for client interface?
Does any of them provide FULL admin functionality, as in creating
extensions, configuring voice mail etc….?
Many thanks for your assistance.
No virus found in this outgoing message.
Checked by AVG.
Version: 7.5.519 / Virus Database: 269.22.1/1352 - Release Date: 3/31/2008
10:13 AM
Hi
My 1.3.1 server does not start when I activate the modules auth and auth_db
modules with the following configuration:
loadmodule "auth.so"
loadmodule "auth_db.so"
[...]
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "user_column", "username")
modparam("auth_db", "password_column", "password")
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "db_url", "mysql://openser:openserrw@192.168.1.4
/openser")
modparam("auth_db", "load_credentials", "")
There is no error in the log file but no other module is loaded whenever I
enable auth and auth_db.
Is there anything else I should enable to configure these two modules ?
When I disable the loading of these two modules everything works fine.
Regards,
Pascal