[asterisk-users] Automon reliability issue
Jaap Winius
jwinius at umrk.to
Mon Feb 11 13:22:58 CST 2008
Hi list,
Can someone please explain how to get one touch recording (automon) to
work reliably? I'm using Asterisk 1.4.14 on a Debian etch system. My
current configuration includes the following settings:
In /etc/asterisk/sip.conf:
[2000]
; Siemens Gigaset S675 IP wireless SIP phone.
type=friend
secret=1234
context=phones-j
dtmfmode=rfc2833
qualify=yes
host=dynamic
[3000]
; Siemens Gigaset E455 wireless analog phone
; attached to Linksys PAP2T SIP adapter.
type=friend
secret=4321
context=phones-j
dtmfmode=rfc2833
qualify=yes
host=dynamic
In /etc/asterisk/features.conf:
[featuremap]
automon => *1
In /etc/asterisk/extensions.conf:
[phones-j]
exten => 2000,1,Dial(SIP/2000,,wW)
exten => 3000,1,Dial(SIP/3000,,wW)
When I make or receive a call with either of these extensions, it's
possible to start and stop recording by pressing "*1". However, this
only works if I press the two keys in quick succession; if I'm not
fast enough, all I see is lots of the following console output:
-- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270
-- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270
-- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270
-- Packet2Packet bridging SIP/1000-081cffb0 and SIP/2000-08241270
In other words, unless I'm also monitoring the console, I can never be
sure that Asterisk has actually started or stopped recording a call
after I press these keys. Frequently, I first have to make several
attempts.
Is there some way to get automon to work reliably, or is the Monitor()
function the only thing we can really count on to record calls?
Thanks,
Jaap
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