This is done in the rtp.conf file. You specify the port range with a start and end number. By default the range is 10000 through 20000.
Leif. > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Alexander Simeonidis > Sent: Thursday, May 13, 2004 10:36 PM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] Asterisk, Configuration of SDP in SIP messages > > Hello everybody, > > I'm new to Asterisk and I'm trying to configure the SIP side. > > I use Asterisk under the following configuration: > > SIP Proxy ---- INTERNET ---- | NAT FIREWALL | ---- Asterisk ---- SIP Phone > A > > I'm trying to put a call from SIP Phone A through Asterisk to the SIP > Proxy. I'm able to deliver messages to SIP Proxy. However, I have noticed > that the port used to deliver the audio changes randomly. I would like to > fix that to a specific range of ports so that I can tell to NAT Firewall > to port forward these particalar ports to Asterisk. I have searched on > documentation and the only thing that I found was how to change the SIP > port but not the media port. Has anybody any ideas on how to solve that > problem or where to look for a solution? > > Regards, > > Alex. > > > ________________________________ > > Help STOP spam with the new MSN 8 <http://g.msn.com/8HMAEN/2731??PS=47575> > and get 2 months FREE* > _______________________________________________ Asterisk-Users mailing > list [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or > update options visit: http://lists.digium.com/mailman/listinfo/asterisk- > users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
