May 2006 - App_rpt-users - lists.keekles.org

[App_rpt] app_rpt.c test version 0-46-6
by Steve Rodgers 06 Jun '06

06 Jun '06

Well, I decided I didn't like the way the autopatch default overrides (patchfarenddisconnect, patchnoct, patchdialtime) were implemented in app_rpt.pre-0-46-4.c so I ripped them out and started over. This method provides much more flexibility. Instead of specifying settings in the node configuration block, you now override the defaults in the function block for autopatch up. In other words, autopatchup can can optionally take comma delimited setting=value pairs. There are currently 5 setting=value pairs defined in app_rpt.pre-0-46-6.c: context=string Override default context with "string" dialtime=ms Specify the max number of milliseconds between phone number ) farenddisconnect=1 Automatically disconnect when called party hangs up noct=1 Don't send repeater courtesy tone during autopatch calls quiet=1 Don't send dial tone, or connect messages. Do not send patch down message when called party hangs up Example: 123=autopatchup,dialtime=20000,noct=1,farenddisconnect=1 The example will set a 20 second limit on dial time between phone number digits, not send the courtesy tone during the time the patch is up, and disconnect the patch if the called party hangs up. By redoing this, we gain a whole bunch more flexibility because we can define several different DTMF sequences for autopatchup which can direct the autopatch to act differently. By defining 3 different patch up sequences, you can now for example, have one for emergency numbers, one for normal numbers, and one for system information messages. You can set the individual autopatch up sequences to use different contexts which can then be added to extensions.conf. Note: I did not remove the autopatch up and down messages from the telemetry block. These are still there so that you can override the default up and down messages. app_rpt.pre-0-46-6c can be downloaded from http://zapatatelephony.com using the following URL: http://zapatatelephony.com/app_rpt.pre-0-46-6.c Note: are running asterisk 1.2.7.1 or earlier, you will need to uncomment the #define for OLD_ASTERISK at the top of the file. Steve WA6ZFT

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[App_rpt] Fwd: app_rpt.c test version 0-46-6: Errata
by Steve Rodgers 14 May '06

14 May '06

Error in the URL: should be www.zapatatelephony.org Steve WA6ZFT ---------- Forwarded Message ---------- Subject: [App_rpt] app_rpt.c test version 0-46-6 Date: Saturday 13 May 2006 21:08 From: Steve Rodgers <hwstar(a)rodgers.sdcoxmail.com> To: app_rpt(a)lists.illiana.net Well, I decided I didn't like the way the autopatch default overrides (patchfarenddisconnect, patchnoct, patchdialtime) were implemented in app_rpt.pre-0-46-4.c so I ripped them out and started over. This method provides much more flexibility. Instead of specifying settings in the node configuration block, you now override the defaults in the function block for autopatch up. In other words, autopatchup can can optionally take comma delimited setting=value pairs. There are currently 5 setting=value pairs defined in app_rpt.pre-0-46-6.c: context=string Override default context with "string" dialtime=ms Specify the max number of milliseconds between phone number ) farenddisconnect=1 Automatically disconnect when called party hangs up noct=1 Don't send repeater courtesy tone during autopatch calls quiet=1 Don't send dial tone, or connect messages. Do not send patch down message when called party hangs up Example: 123=autopatchup,dialtime=20000,noct=1,farenddisconnect=1 The example will set a 20 second limit on dial time between phone number digits, not send the courtesy tone during the time the patch is up, and disconnect the patch if the called party hangs up. By redoing this, we gain a whole bunch more flexibility because we can define several different DTMF sequences for autopatchup which can direct the autopatch to act differently. By defining 3 different patch up sequences, you can now for example, have one for emergency numbers, one for normal numbers, and one for system information messages. You can set the individual autopatch up sequences to use different contexts which can then be added to extensions.conf. Note: I did not remove the autopatch up and down messages from the telemetry block. These are still there so that you can override the default up and down messages. app_rpt.pre-0-46-6c can be downloaded from http://zapatatelephony.com using the following URL: http://zapatatelephony.com/app_rpt.pre-0-46-6.c Note: are running asterisk 1.2.7.1 or earlier, you will need to uncomment the #define for OLD_ASTERISK at the top of the file. Steve WA6ZFT _______________________________________________ App_rpt mailing list App_rpt(a)lists.illiana.net http://lists.illiana.net/mailman/listinfo/app_rpt -------------------------------------------------------

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Re: [App_rpt] "How to" for a simplex radio station
by Steve Rodgers 09 May '06

09 May '06

Francois, We support simplex operation with the duplex= statement in the node block. Normally the default is duplex=2 which is a full duplex repeater. Duplex=0 is a simplex node with no telemetry, and duplex=1 is a simplex node with telemetry. Telemetry is command responses, ID's, etc. For node use, app_rpt requires 2 FXS ports are required per node (one for TX and one for RX) Regardless if it is duplex or not. On Monday 08 May 2006 05:42, f6hqz-m(a)hamwlan.net wrote: > Hello Jim and Steve, > > I hope all is fine for you two :-) > > I am tempting to connect few simplex stations (1 VHF, 1 UHF and 1 HF) to my > Asterisk in my chack. > I want to connect them directly to my TDM2433E card, because I have "too > much" FXS lines available. > > So, I am wondering how to use this FXS lines and I have probably missed > something because I can't. > I have searched some examples on Voi-info, Zapata and QRV web sites, > without seen any kind of help for my usage or few explanations about the > method used for COR and PTT actions through such a FXS. You need our ARIB board for this. > > If I read the zapata. and zaptel.conf exampel files included into the > Asterisk distro, I can see that it seems to be possible to use 2 different > kinds of signaling for FXS and radio : E&M or "sf" or "fxsks" or "fxsgs" in > zaptel.conf, and em_txrx or sf_txrx in zapata.conf... > I don't understand well what I must use for my situation (I don't know how > works E&M signaling exactly). > I have choosen em_txrx thinking that this could be used on only one FXS > channel to have COR and PTT. E&M is just a placeholder for implementing COR and PTT signalling and only used with the quad radio PCI card. You do it like this... zaptel.conf: fxogs=17 # TX bench test pair channel 17 on channelbank fxols=18 # RX bench test pair channel 18 on channelbank zapata.conf: ; Channel 17,18 callwaiting=no threewaycalling=no transfer=no context=ext-disconnected mailbox= callerid="Bench TX"<117> signalling=fxo_tx channel = 17 signalling=fxo_rx context=ext-disconnected threewaycalling=no callwaiting=no transfer=no mailbox= callerid="Bench RX"<118> channel = 18 rpt.conf: [112] ; node block rxchannel = Zap/18 ; Rx audio/signalling channel ; Note: if you use a unified interface (tx/rx on one channel), only ; specify the rxchannel and the txchannel will be assumed from the rxchannel txchannel = Zap/17 ; Tx audio/signalling channel duplex = 1; Simplex node with telemetry ; Other config stuff for the node block (don't assume that the lines above are all that is required!!!! . . . > > But, I can't access to my "node" when I tempt to call it with apt(100) or > apt(100|X) into extensions.conf. > Asterisk console tells me : > May 8 14:32:23 WARNING[1562]: app_rpt.c:5866 rpt_exec: Cannot find > specified system node 001 > > In my rpt.conf, I have very few lines for this first test link : > > [001] ; Node ID of first repeater > ;VHF Base #1 FT290R > rxchannel = Zap/2 ; Rx audio/signalling channel > context = HAMWLAN ; dialing context for phone > callerid = ""F6HQZ_VHF_BASE1" <77005> ; Callerid for phone calls > > [nodes] > 001=radio(a)127.0.0.1/001,NONE > > So dear OM's, could you help me to start this simplex stations that I want > to add to your main Asterisk radio network ? > May be it could be nice to write some such paper in the voip-info wiki to > have a real example to start easyly ? > > I hope to start my radio link before the may 20th, due to a HAMfest in > France with hudreds of HAMs. > I will be pleased to annouce and show this first France/USA radio link with > Asterisk boxes and your RPT application working. > > For now, I have (too) few HAM's connected to my Asterisk box, chating > together, but having only receivers access (2 HF RX on 7 MHz band). > > If you want check my HF listening channels, I can give you an IAX2 account > or we can create an IAX2 trunk immediatly that could be used later for our > future radio link between our two countries. > > Best 73's, > F6HQZ, > Francois BERGERET, > France.

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[App_rpt] Fwd: Re: Fwd: app_rpt.pre-046-4.c: errata
by Steve Rodgers 09 May '06

09 May '06

---------- Forwarded Message ---------- Subject: Re: [App_rpt] Fwd: app_rpt.pre-046-4.c: errata Date: Monday 08 May 2006 20:54 From: Steve Rodgers <hwstar(a)rodgers.sdcoxmail.com> To: Shane Young KØSEY <apprpt(a)shaneyoung.com> Like this... [functions] . . . 5=macro,1 ; assigns *5 as the macro lead in sequence. . . . [macro] 1 = *32011# 2 = *12001*12011*12043# 3 = 32050*32060# Now, dialing *51 executes *32011, *52 executes *12001*12011*12043 Well you get the picture. Don't forget the trailing # in the macro def, as this is the termination character. Steve, WA6ZFT On Monday 08 May 2006 17:10, you wrote: > It's working now and it works great! > > Now, how do I get those macros to work? > > I have > [macro] > 11 = *66128217391 > > which, as I understand, if I hit "11" should have the same function as > dialing *6+612-821-7391 > > Quoting Steve Rodgers <hwstar(a)rodgers.sdcoxmail.com>: > > Shane > > > > patchup and patchdown belong in the telemetry block, not the node block. > > Where is your telemetry block? Also, I don't see a telemetry=telemetry > > assignment in your node block stanza. Did you delete the telemetry block > > stanza and just use the built-in defaults? > > > > Steve WA6ZFT > > > > On Sunday 07 May 2006 20:44, you wrote: > > > 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(a)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(a)rodgers.sdcoxmail.com> > > > > To: Asterisk Repeater Controler <app_rpt(a)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 > > > > > > > > ------------------------------------------------------- > > > > _______________________________________________ > > > > App_rpt mailing list > > > > App_rpt(a)lists.illiana.net > > > > http://lists.illiana.net/mailman/listinfo/app_rpt > > > > > > ---------------------------------------------------------------- > > > This message was sent using IMP, the Internet Messaging Program. > > ---------------------------------------------------------------- > This message was sent using IMP, the Internet Messaging Program. -------------------------------------------------------

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[App_rpt] Fwd: app_rpt.pre-046-4.c: errata
by Steve Rodgers 07 May '06

07 May '06

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(a)rodgers.sdcoxmail.com> To: Asterisk Repeater Controler <app_rpt(a)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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[App_rpt] app_rpt.pre-046-4.c
by Steve Rodgers 04 May '06

04 May '06

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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