here is my report of possible rtp/nat and playback bug:
sip.conf contains:
[general]
nat=yes
canreinvite=nonat
; canreinvite=no
qualify=yes
and then two dynamic user entries
The prblem is as follows. When Asterisk attempts to play, it sends its first packet
to the wrong place.
-- Executing [886@phones:1] Answer("SIP/dave-09917180", "") in new stack
-- Executing [886@phones:2] Playback("SIP/dave-09917180", "/tmp/asterisk") in new stack
Sent RTP packet to XXX.247.174.254:17498 (type 00, seq 010488, ts 000160, len 000160)
-- <SIP/dave-09917180> Playing '/tmp/asterisk' (language 'en')
Got RTP packet from XXX.247.174.254:5441 (type 00, seq 003241, ts 3069300, len 000160)
... thousands of these
with 1.2, the first (and remaining) packets all go to the same place that the received
packets come from.