I'm currently running AstLinux 1.0.2 (thank you team, flawless upgrade!) with
Asterisk 1.8.x. My SIP provider allows for up to 5 concurrent calls; I had
limited my port range in both rtp.conf and my firewall to 10000 - 10020.
However, I recently noticed that one of my 3 SIP extensions stopped ringing.
Looking at the CLI, I see the following error:
> [Feb 28 19:46:54] WARNING[6396]: app_dial.c:2218 dial_exec_full: Unable to
> create channel of type 'SIP' (cause 20 - Unknown)
> [Feb 28 19:46:54] ERROR[6396]: res_rtp_asterisk.c:511 ast_rtp_new: Oh dear...
> we couldn't allocate a port for RTP instance '0x8885d98'
When I doubled the number of RTP ports available in rtp.conf, all extensions
rang as expected.
I'm not implying this is an AstLinux issue, but am hoping this community can
help me better understand how many RTP ports could be consumed? My goal is to
limit the number of open ports on my firewall.
many thanks,
Shamus
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