There is a buch of commands you can use for uri maniputlation: strip, prefix, subst_uri.
-jiri
At 15:52 25/07/2006, Robert Zorop wrote:
>Hi guys.
> I'd like to hear about different methods of implementing cisco like transtation rules in SER. I was thinking about AVPs, but still couldn't find the ight way.
> Is someone is doing it inside or outside ser, please shed a light on this!. What i need is to stip some numbers, and then add some other (variables) before proxying the calls.
>
>
>Thanks in advance.
>
>_______________________________________________
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>Serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hello all,
I'm trying to implement routing in general and I'm also trying to use
the lcr module. I'm a little confused as to where in the picture does
xml fit in.
While reading through documentation I came across the acronym RPID which
seems to be an implementation of routing information in an xml document,
that's passed to lcr module as a parameter. What are the mechanics
behind this?
Thanks in advance for your help,
Marvin
All,
I've just finished my first release of OpenSER Administrator, a web interface
for managing OpenSER. It's GPL'd and written in Ruby on Ralis, and can be
downloaded from openseradmin.sourceforge.net
Please let me know what you think, and let me know of any features you want,
bugs you find, etc.
Thanks,
Mike Williams
Hi,
I'm deploying a SER + Asterisk architecture, where SER is used to manage
acc, users database and sip routing, and Asterisk is used for voicemail
and PSTN gateway.
The system is already able to make and receive calls from the PSTN,
although, only after the call has been established it can be hung up
with success; when it is still ringing, if any side hungs up the call,
it still keeps ringing on the other side. Observing with Ethereal, we
concluded that in this erroneous cases, the CANCEL SIP request isn't
transmitted from the SER to Asterisk (if cancelled from the VoIP side)
being transmitted a "404 User Not Found" message from SER to Sip Phone.
If hung from the PSTN side, the sip phone keeps calling after that, and
ends calling by time-out being observed a "486 Busy Here" status message
from Asterisk to SER and then from SER to sip phone.
Any help, please?
Regards,
Ricardo.
Wow! Thanx!!!
On 8/7/06, sip <sip(a)arcdiv.com> wrote:
> Usually, that's done in the /etc/my.cnf file. Is it not working there?
>
> My my.cnf file has a client ([client]) and server ([mysqld]) section.
>
> My server section (the non-commented part) looks like:
>
> [mysqld]
> port = 3306
> socket = /var/lib/mysql/mysql.sock
>
> The client section looks the same (since I like the defaults to work the same
> when connecting via the command-line client):
>
> [client]
> [mysqld]
> port = 3306
> socket = /var/lib/mysql/mysql.sock
>
>
> There were a few packages I had that didn't come with an /etc/my.cnf file but
> still read from one if it was there.
>
> N.
>
> On Mon, 7 Aug 2006 19:47:54 +0700, Andrey Kouprianov wrote
> > Hi,
> >
> > I know it's the wrong list and I dont even expect anyone to answer.
> > Anyway... Does anyone know how to specify different listening TCP
> > port and socket name when installing mysql server from FreeBSD ports?
> >
> > Andrey.
> > _______________________________________________
> > Serusers mailing list
> > Serusers(a)lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
Hi,
I know it's the wrong list and I dont even expect anyone to answer.
Anyway... Does anyone know how to specify different listening TCP port
and socket name when installing mysql server from FreeBSD ports?
Andrey.
Hi all
I am new to openser and would like to configure my routes. I would like to
have some samples. Please point me on the right direction.
Regards
Lens Oreste
