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)
when 9002 call 9001 finish,no call found in “/var/spool/asterisk/monitor”, why?
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.
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
////////////////////////////////////////
what about these ?
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?