[Asterisk-Users] Asterisk, Configuration of SDP in SIP messages

[Leif Madsen]

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: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> admin at lists.digium.com] On Behalf Of Alexander Simeonidis
> Sent: Thursday, May 13, 2004 10:36 PM
> To: asterisk-users at lists.digium.com
> 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 Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or
> update options visit: http://lists.digium.com/mailman/listinfo/asterisk-
> users