[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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