asterisk users - Jun 2006 - Custom Extension halting execution upon caller hanging up

Hello, list!

I'm having some trouble with A@H 2.7(?), Asterisk 1.2.5, inasmuch as 
my custom extension is not continuing execution when the caller hangs 
up. (Please excuse the sterilized output.)

Here's how it's supposed to go:

exten => 2,8,Monitor(wav,${TIMESTAMP})
exten => 2,9,Dial(SIP/Provider/8005551212)
exten => 2,10,Macro(record-cleanup)

If the caller hangs up before the callee does, execution of the 
custom extension halts and does not continue to priority 10 
(record-cleanup), where sox is used to reverse the audio files and 
then mix them then reverse them again so they'll be in sync (since 
inbound audio only starts from call-answered but outbound audio 
starts from the beginning of ringback).

Asterisk provides this debug output to the console (internal 
extension 101 is the caller):
     -- Called Provider/8005551212
     -- SIP/Provider-993d is making progress passing it to SIP/101-1666
     -- SIP/Provider-993d answered SIP/101-1666

The call proceeds normally, but then Asterisk spits this out the 
moment the caller hangs up first:
   == Spawn extension (custom-extension, 2, 9) exited non-zero on 
'SIP/101-1666'

How can I prevent the extension from bailing before I have a chance 
to clean up the recording?

Thanks in advance!

--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada

Hello, list!

I'm having some trouble with A@H 2.7(?), Asterisk 1.2.5, inasmuch as 
my custom extension is not continuing execution when the caller hangs 
up. (Please excuse the sterilized output.)

Here's how it's supposed to go:

exten => 2,8,Monitor(wav,${TIMESTAMP})
exten => 2,9,Dial(SIP/Provider/8005551212)
exten => 2,10,Macro(record-cleanup)

If the caller hangs up before the callee does, execution of the 
custom extension halts and does not continue to priority 10 
(record-cleanup), where sox is used to reverse the audio files and 
then mix them then reverse them again so they'll be in sync (since 
inbound audio only starts from call-answered but outbound audio 
starts from the beginning of ringback).

Asterisk provides this debug output to the console (internal 
extension 101 is the caller):
     -- Called Provider/8005551212
     -- SIP/Provider-993d is making progress passing it to SIP/101-1666
     -- SIP/Provider-993d answered SIP/101-1666

The call proceeds normally, but then Asterisk spits this out the 
moment the caller hangs up first:
   == Spawn extension (custom-extension, 2, 9) exited non-zero on 
'SIP/101-1666'

How can I prevent the extension from bailing before I have a chance 
to clean up the recording in priority 10?

Thanks in advance!

--
Alexander Burke, A+, CCNA
Kingston, Ontario, Canada