help! about automon

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help! about automon

Postby bashan » Tue Feb 06, 2007 2:26 am

1)Asterisk:1.4.0
2)extensions.conf
[globals]
DYNAMIC_FEATURES=>automon
[tutorial]
exten =>9001,1,Dial(SIP/9001,,wW)
exten =>9002,1,Dial(SIP/9002,,wW)

3) features.conf
[featuremap]
automon => *1

when 9002 call 9001 finish,no call found in "/var/spool/asterisk/monitor", why?
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Postby baconbuttie » Tue Feb 06, 2007 2:29 am

did either party initiate the recording by pressing '*1' ? did the CLI report that it was detected ? the CLI (or log) will probably also tell you where it's been stored too.
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Postby bashan » Tue Feb 06, 2007 2:43 am

thank you for your help!
below is the cli log
////////////////////////////////////////////////////////////
Registered SIP '9001' at 192.168.8.1 port 16664 expires 120
-- Saved useragent "RTC/1.2.4949 (BOL SIP Phone 1005)" for peer 9001
-- Executing [9001@tutorial:1] Dial("SIP/9002-082378e8", "SIP/9001||wW") in new stack
-- Called 9001
-- SIP/9001-08241478 is ringing
-- SIP/9001-08241478 answered SIP/9002-082378e8
-- Packet2Packet bridging SIP/9002-082378e8 and SIP/9001-08241478
== Spawn extension (tutorial, 9001, 1) exited non-zero on 'SIP/9002-082378e8'
-- Unregistered SIP '9001'
-- Registered SIP '9001' at 192.168.8.1 port 16664 expires 120
-- Saved useragent "RTC/1.2.4949 (BOL SIP Phone 1005)" for peer 9001
////////////////////////////////////////
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Postby baconbuttie » Tue Feb 06, 2007 2:58 am

baconbuttie wrote:did either party initiate the recording by pressing '*1' ? did the CLI report that it was detected ? the CLI (or log) will probably also tell you where it's been stored too.


what about these ?
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Postby wharfrat » Wed Feb 21, 2007 4:03 pm

I had the same or similar problem with the stable release of 1.4. I ended up having to specify canreinvite=no in my sip.conf so that Asterisk would not drop out of the call path and thus "hear" *1 and start recording.

From what I gather, having 'w' or 'W' in the DIAL statement is supposed to let Asterisk know not to do this, but I could not get automon (or any other dynamic features) working for me until I specified 'canreinvite=no' I am currently testing * and all my extensions are on the same subnet (for now.) It's my understanding that when start taking/making calls outside of my local network, this will not be an issue.

Speaking of which, does anybody know if there is a way to specify this globally in sip.conf, rather than per extension?
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