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@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@rodgers.sdcoxmail.com> To: Asterisk Repeater Controler <app_rpt@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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