[asterisk-users] [1.4] Failed callfile doesn't jumpto"failed"extension

[Gilles]

On Fri, 18 Mar 2011 16:48:28 -0700 (PDT), Steve Edwards
<asterisk.org at sedwards.com> wrote:
>Somehow, I'm guessing that 'failed' means that something failed while 
>processing the call file or that the call failed to answer, not that 
>somebody terminated the call.

Thanks guys. After testing with a PCI card + Dahdi, and then with a
Linksys 3102, turns out that neither jumps to the "failed" or "h"
extension when the remote number is busy, ie. already engaged (with no
support for callwaiting, ie. two-way calling)

========== extensions.conf
[internal]
;call from XLite
;exten => _5.,1,Dial(Dahdi/1/${EXTEN})
exten => _5.,1,Dial(SIP/3102-fxo/${EXTEN})

exten => h,1,NoOp(Called ended with ${DIALSTATUS})

exten => failed,1,NoOp(Call ended with ${REASON})
========== CLI
== Using SIP RTP CoS mark 5
-- Executing [5551234 at internal:1] Dial("SIP/xlite-0000000e",
"SIP/3102-fxo/5551234") in new stack
== Using SIP RTP CoS mark 5
-- Called 3102-fxo/5551234

#Here, phone is still ringing, but Asterisk wrongly says it has
"answered"
-- SIP/3102-fxo-0000000f is ringing
-- SIP/3102-fxo-0000000f answered SIP/xlite-0000000e

#Says it bridged calls although remote end hasn't answered
-- Packet2Packet bridging SIP/xlite-0000000e and SIP/3102-fxo-0000000f
========== 

As I no longer have a "real" landline, it could be due to the way my
ADSL VoIP landline works. Bottom line: I can't use that line to write
a robocall.

Thanks guys.