I’m having a weird problem with Asterisk 1.2.13, running on a Linux box.
I have a firewall, which lets any needed port through the ports 30000…32000 for RTP, as specified in rtp.conf:
[general]
rtpstart=30000
rtpend=32000
BUT: the RTP stream is often sent through a port which is not in this range!!!
So there is often no audio!
Thanks for your post! Unfortunately there is no device on the other end of the SIP connection but a SIP provider: www.voipdiscount.com
Can I still force the use of a specified port range somehow?
(I don’t think voipdiscount will change his settings, just for me ! )
Thanks, I’ll for sure ask them. Too bad, until know they seem to use just a random port from port 1024 upwards.
Maybe someone knows a possibility to enforce the use of a port range?
I know if they are using a sip aware firewall it will probably not work, but maybe they have just left any port in a wide range open. (Which is probably quite uncommon.)
Or I should just use a STUN enabled SIP proxy in front of Asterisk?
GRR… Things getting complicated.