[App_rpt] Fwd: app_rpt.pre-046-4.c: errata

[Shane Young KØSEY]

I'm using the following config and I am unable to change the patchup and patchdown messages.  the
patchfarenddisconnect *is* working.


[2040]
rxchannel = Zap/31
txchannel = Zap/30
functions = functions-repeater
link_functions = functions-different
tonezone = us
;context = cos1
;context=repeater-out
context=sip-phones
callerid = "Repeater" <6128217300>
idrecording = rpt/idk0sey
accountcode=RADIO
funcchar = *
endchar = #
patchfarenddisconnect=1
patchup = silence/1
patchdown = local/chimes/1_tone_chime
hangtime=1000
totime=180000
idtime=300000
politeid=30000
idtalkover=|ik0sey
unlinkedct=ct3
tailmessagetime=360000
tailsquashtime=  90000
tailmessagelist=local/current-temp
wait_times=wait-times


Quoting Steve Rodgers <hwstar at rodgers.sdcoxmail.com>:

>
> Errata:
>
> Replace all instances of farenddisconnect with patchfarenddisconnect in my
> prior message.
>
> Steve WA6ZFT
>
> ----------  Forwarded Message  ----------
>
> Subject: app_rpt.pre-046-4.c
> Date: Wednesday 03 May 2006 21:49
> From: Steve Rodgers <hwstar at rodgers.sdcoxmail.com>
> To: Asterisk Repeater Controler <app_rpt at lists.illiana.net>
>
> There is a pre release (experimental) version of app_rpt.c at:
>
> http://zapatatelephony.org/app_rpt.pre-046-4.c
>
> This version supports the following additional features:
>
> 1. Hot reload of rpt.conf from the asterisk command line. Typing "rpt reload"
> will reload rpt.conf without causing a restart of the application and without
> disconnecting any links. Note: changing node number stanzas ( n square
> brackets) will still require that app_rpt be restarted.
>
> 2. Restart of app_rpt from the command line without having to restart
> asterisk. Typing "rpt restart" will restart app_rpt, but leave the rest of
> asterisk running.
>
> 3. Far end disconnect support on the autopatch with the patchfarenddisconnect
> setting in the node configuration block. if farenddisconnect is set to 1
> (farenddisconnect=1) then when the called party hangs up, the call terminated
> message will be played. Setting farenddisconnect to 0 keeps the original
> default behavior of going to congestion if the called party hangs up.
>
>
> 4. Automatic disconnect during dialing if a radio user goes out of range. The
> patchdialtime setting in the node configuration block does this. This value
> is in milliseconds. For example a 20 second timer would be configured by
> setting patchdialtime to 20000 (patchdialtime=20000). This timer resets when
> ever a dtmf digit is dialed, and is also used to limit the time the
> congestion tone is played if asterisk declares there is no route to a number.
> The default is patchdialtime=0 which disables the dial timer.
>
>
> 5. The patchnoct setting in the node configuration block can be used to
> suppress the CT during an autopatch. Setting patchnoct to 1 (patchnoct=1)
> will suppress the courtesy tone during an autopatch, and setting patchnoct to
> 0 will keep the default behaviour. The default is patchnoct=0.
>
>
> 6. Override of the autopatch connect message. The patchup setting in the
> telemetry block allows the user to specify a file, morse message or tone
> sequence instead of the default message of "connecting". The syntax is the
> same used by all of the other entries in the telemetry block. The default is
> patchup=rpt/callproceeding.
>
>
> 7. Override the autopatch disconnect message. The patchdown setting in the
> telemetry block, allows the user to specify a file, morse message or tone
> sequence instead of the default message "call terminated". The syntax is the
> same used by all of the other entries in the telemetry block. The default is:
> patchdown=rpt/callterminated
>
>
> If you are running a tarball release of asterisk version 1.27.1 or earlier,
> you will need to uncomment the "OLD_ASTERISK" line at the top of app_rpt.c
> This is because the SVN version has undergone a lot of change in relation to
> the stable tarballs.
>
>
> A note of caution: This is test code, use at your own risk.
>
> We would be delighted if others tested it for us and reported any problems
> with it before we release it on Digium SVN.
>
>
>
> Steve Rodgers,
> WA6ZFT
>
> -------------------------------------------------------
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>




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