asterisk users - Mar 2008 - Newbie One-touch Recording: Does not work

I thought it was quite easy to implement but I cannot get one-touch
recording to work.  Here are the changes what I did:

I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.

Any suggestions?

Here is the console log:

-- Accepting call from '298741000' to '98553166' on channel
0/10, span 1
-- Executing [98553166 at default:1] Macro("Zap/10-1",
"stdexten|5166|5166") in new stack
-- Executing [s at macro-stdexten:1] SetMusicOnHold("Zap/10-1",
"cpwr") in
new stack
-- Executing [s at macro-stdexten:2] Set("Zap/10-1",
"DYNAMIC_FEATURES=automon") in new stack
-- Executing [s at macro-stdexten:3] Dial("Zap/10-1",
"SIP/5166|20|wW") in
new stack
-- Called 5166
-- SIP/5166-0930f2d0 is ringing
-- SIP/5166-0930f2d0 answered Zap/10-1
== Spawn extension (macro-stdexten, s, 3) exited non-zero on 'Zap/10-1'
in macro 'stdexten'
== Spawn extension (macro-stdexten, s, 3) exited non-zero on 'Zap/10-1'
-- Hungup 'Zap/10-1'

extensions.conf

[macro-stdexten];
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
exten => s,1,SetMusicOnHold(cpwr)
exten => s,2,Set(DYNAMIC_FEATURES=automon)      
exten => s,3,Dial(SIP/${ARG2},20,wW)
exten => s,4,Goto(s-${DIALSTATUS},1)            
exten => s-NOANSWER,1,Voicemail(${ARG1},u)
exten => s-NOANSWER,2,Goto(default,s,1)
exten => s-BUSY,1,Voicemail(${ARG1},b)
exten => s-BUSY,2,Goto(default,s,1)
exten => _s-.,1,Goto(s-NOANSWER,1)              
exten => a,1,VoicemailMain(${ARG1})

features.conf

[general]
parkext => 700
parkpos => 701-720
context => parkedcalls
courtesytone = beep
parkedplay = both
parkedmusicclass=cpwr

[featuremap]
automon => *1

Does 'show features' display the correct information?

PaulH


On Thu, 2008-03-13 at 15:39 +1100, Lee, John (Sydney)
wrote:
> I thought it was quite easy to implement but I cannot get one-touch
> recording to work.  Here are the changes what I did:
> 
> I restarted Asterisk after the change (because reload does not work for
> changes in features.conf).
> I press *1 on the Polycom IP600 phone to record a conversation but no
> new wav file appear in /var/spool/asterisk/monitor or elsewhere.
> 
> Any suggestions?
> 
> Here is the console log:
> 
> -- Accepting call from '298741000' to '98553166' on channel
0/10, span 1
> -- Executing [98553166 at default:1] Macro("Zap/10-1",
> "stdexten|5166|5166") in new stack
> -- Executing [s at macro-stdexten:1] SetMusicOnHold("Zap/10-1",
"cpwr") in
> new stack
> -- Executing [s at macro-stdexten:2] Set("Zap/10-1",
> "DYNAMIC_FEATURES=automon") in new stack
> -- Executing [s at macro-stdexten:3] Dial("Zap/10-1",
"SIP/5166|20|wW") in
> new stack
> -- Called 5166
> -- SIP/5166-0930f2d0 is ringing
> -- SIP/5166-0930f2d0 answered Zap/10-1
> == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'Zap/10-1'
> in macro 'stdexten'
> == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'Zap/10-1'
> -- Hungup 'Zap/10-1'
> 
> extensions.conf
> 
> [macro-stdexten];
> ; Standard extension macro:
> ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
> ;   ${ARG2} - Device(s) to ring
> exten => s,1,SetMusicOnHold(cpwr)
> exten => s,2,Set(DYNAMIC_FEATURES=automon)      
> exten => s,3,Dial(SIP/${ARG2},20,wW)
> exten => s,4,Goto(s-${DIALSTATUS},1)            
> exten => s-NOANSWER,1,Voicemail(${ARG1},u)
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(${ARG1},b)
> exten => s-BUSY,2,Goto(default,s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)              
> exten => a,1,VoicemailMain(${ARG1})
> 
> features.conf
> 
> [general]
> parkext => 700
> parkpos => 701-720
> context => parkedcalls
> courtesytone = beep
> parkedplay = both
> parkedmusicclass=cpwr
> 
> [featuremap]
> automon => *1
> 
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

On our system i got:

Zap/1-1 answered SIP/106-091a2750
    -- User hit '*1' to record call. filename: wav|
auto-1205385048-106-0434225491|m

Our dialplan looks like:

_0XXXXXXXXX' =>  1. Dial(zap/g1/${EXTEN}||Ww)                 

(from show dialplan)

PaulH


On Thu, 2008-03-13 at 15:39 +1100, Lee, John (Sydney)
wrote:
> I thought it was quite easy to implement but I cannot get one-touch
> recording to work.  Here are the changes what I did:
> 
> I restarted Asterisk after the change (because reload does not work for
> changes in features.conf).
> I press *1 on the Polycom IP600 phone to record a conversation but no
> new wav file appear in /var/spool/asterisk/monitor or elsewhere.
> 
> Any suggestions?
> 
> Here is the console log:
> 
> -- Accepting call from '298741000' to '98553166' on channel
0/10, span 1
> -- Executing [98553166 at default:1] Macro("Zap/10-1",
> "stdexten|5166|5166") in new stack
> -- Executing [s at macro-stdexten:1] SetMusicOnHold("Zap/10-1",
"cpwr") in
> new stack
> -- Executing [s at macro-stdexten:2] Set("Zap/10-1",
> "DYNAMIC_FEATURES=automon") in new stack
> -- Executing [s at macro-stdexten:3] Dial("Zap/10-1",
"SIP/5166|20|wW") in
> new stack
> -- Called 5166
> -- SIP/5166-0930f2d0 is ringing
> -- SIP/5166-0930f2d0 answered Zap/10-1
> == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'Zap/10-1'
> in macro 'stdexten'
> == Spawn extension (macro-stdexten, s, 3) exited non-zero on
'Zap/10-1'
> -- Hungup 'Zap/10-1'
> 
> extensions.conf
> 
> [macro-stdexten];
> ; Standard extension macro:
> ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
> ;   ${ARG2} - Device(s) to ring
> exten => s,1,SetMusicOnHold(cpwr)
> exten => s,2,Set(DYNAMIC_FEATURES=automon)      
> exten => s,3,Dial(SIP/${ARG2},20,wW)
> exten => s,4,Goto(s-${DIALSTATUS},1)            
> exten => s-NOANSWER,1,Voicemail(${ARG1},u)
> exten => s-NOANSWER,2,Goto(default,s,1)
> exten => s-BUSY,1,Voicemail(${ARG1},b)
> exten => s-BUSY,2,Goto(default,s,1)
> exten => _s-.,1,Goto(s-NOANSWER,1)              
> exten => a,1,VoicemailMain(${ARG1})
> 
> features.conf
> 
> [general]
> parkext => 700
> parkpos => 701-720
> context => parkedcalls
> courtesytone = beep
> parkedplay = both
> parkedmusicclass=cpwr
> 
> [featuremap]
> automon => *1
> 
> 
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

My guess (from your features) is that the * for disconnect and *1 for
records are clashing - maybe set disconnect to **73 to avoid this.

And - yes, it can be tuned:

;featuredigittimeout = 500      ; Max time (ms) between digits for
                                ; feature activation  (default is 500ms)

later,

PaulH


On Thu, 2008-03-13 at 17:22 +1100, Lee, John (Sydney)
wrote:
> > I think the problem is "*1" is being ignored or cannot be
transmitted
> > successfully to Asterisk.
> 
> Finally I resolved the problem.
> For some reasons, the "*" and "1" must be pressed
pretty quickly
> together on the Polycom phone before it can be transmitted successfully
> to Asterisk.
> I think I cannot deny that I am getting older and older (or more mature
> :-)
> 
> Does anyone know if that can be tuned?
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Just chatted to one of the guys here - he said that if you set it to a
single DTMF digit, it can be pretty good...anything more, not so
reliable.

PaulH


On Thu, 2008-03-13 at 17:22 +1100, Lee, John (Sydney)
wrote:
> > I think the problem is "*1" is being ignored or cannot be
transmitted
> > successfully to Asterisk.
> 
> Finally I resolved the problem.
> For some reasons, the "*" and "1" must be pressed
pretty quickly
> together on the Polycom phone before it can be transmitted successfully
> to Asterisk.
> I think I cannot deny that I am getting older and older (or more mature
> :-)
> 
> Does anyone know if that can be tuned?
> 
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

----- Original Message -----
From: "John Lee (Sydney)" <John.Lee at compuware.com>
>Does anyone know if that can be tuned?
Sure... go to features.conf, and change the value of the featuredigittimeout
option.

---
Jared Smith
Community Relations Manager
Digium, Inc.