[asterisk-users] SIP RTP ports not released when channel is hung up

[Armin Schindler]

Hello,

using Asterisk 1.4.28, I encountered a problem with SIP
RTP port allocation.

I found some entries in mailinglist and bugtracker regarding
this issue, but only old ones.

My rtp.conf has
  [general]
  rtpstart=30000
  rtpend=30100

so 100 ports available. I know that up to 4 ports per channel can be used
and so up to 25 channels are possible.
But even earlier I often get the error about "No RTP ports remaining".

I had a look at
  netstat -nuap
and it shows that a lot of ports are still assigned, even if there is no
channel in use.
But "sip show channels" show a lot of (unused) entries with no
codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.

Why aren't RTP ports released when not in use?

Or is there a possibility to configure this behaviour?

Thanks,
Armin