[VoiceOps] Asterisk 1.8.6.0 unable to allocate a port for RTP instance

[Joseph Jackson]

Here ya go.  Its a very basic configuration.  We basically just use
this server as a sip/media proxy for some call center agents so we can
record the audio from a central location.  The calls are generated
from a shoretel system which sends the SIP calls to the asterisk box
then the asterisk box sends everything on to the carrier sip address.
The agents are around 30 at a time.  They do outbound calls and then
conferencing.

Thanks.

;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000
;
; Whether to enable or disable UDP checksums on RTP traffic
;
;rtpchecksums=no
;
; The amount of time a DTMF digit with no 'end' marker should be
; allowed to continue (in 'samples', 1/8000 of a second)
;
;dtmftimeout=3000
; rtcpinterval = 5000 	; Milliseconds between rtcp reports
			;(min 500, max 60000, default 5000)
;
; Enable strict RTP protection. This will drop RTP packets that
; do not come from the source of the RTP stream. This option is
; disabled by default.
; strictrtp=yes


On Fri, Sep 16, 2011 at 1:05 AM, Alex Balashov
<abalashov at evaristesys.com> wrote:
> Just out of curiosity, what's your /etc/asterisk/rtp.conf say?
>
> --
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
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