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[App_rpt] progress and stability of asterisk and app_rpt
by Eric Fort 04 Jul '06

04 Jul '06

I am looking at asterisk and app_rpt as a solution for a ship to shore communications platform where safety of life and property may be an issue at times. We will be using dispatch remote from multiple radio sites. Dispatch is now and would continue to be linked to the remotes via VoIP links. at each site there is also phone patch capability for ship to shore phone calls, the main service offered, along with radios for amtor, telex, sitor, and pactor modes for data, email etc. Is asterisk and app_rpt up to the task? bottom line it just has to work. Is it good enough that you would trust it to work if your boat was sinking and you needed a dispatcher to send help before the circling fishes found you tasty (propigation from dispatch to rescue not withstanding)? Eric

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Re: [App_rpt] Interface questions
by hwstar＠rodgers.sdcoxmail.com 02 Jul '06

02 Jul '06

Jon, As Jim stated in a prior E-mail, you must make sure the channel bank supports DPO (Groundstart mode) or the transmitter will stay keyed all the time. The transmitter is keyed by the presence of battery on the TX pair. Steve. > > From: jon(a)unicomsystems.com (Jon Teh) > Date: 2006/06/27 Tue PM 07:13:52 PDT > To: Asterisk Repeater Controler <app_rpt(a)lists.illiana.net> > CC: Jon Teh <jon(a)unicomsystems.com> > Subject: Re: [App_rpt] Interface questions > > On Tue, Jun 27, 2006 at 06:55:14AM -0700, Steve Rodgers wrote: > > > > Jon, > > > > See responses below. > > > > Some questions for you: > > > > Will you be using this in an Amatuer Radio repeater system or a commercial > > system? > > Current plans are not amateur related, so this shouldn't be a problem. > > > > > Last time I checked, Australia does not permit Amateur repeaters connect to > > the phone system (via automatic phone patching or autopatching) is this still > > the case? The reason I ask is that our systems in the US have this > > capability, and if we are linked together, would it be breaking the rules if > > someone in the US did this and and we were linked to a system in Australia? In > > this case, the phone conversation taking place on the US system would be > > rebroadcast on the Australian system. > > > > I know that the Australian Communications and Media Authority put out > a paper specifically stating that Internet linking was permitted and was > not considered connecting to the PSTN. If there's phone patches in the US, > then I guess that makes it a bit grey. But unless it's a major problem, > I doubt ACMA would bother as they generally have higher priority things > to police than amateur radio. > > > > > > > Hrm, they certainly have plenty, and at a good price, but overseas UPS > > >shipping is a killer, about USD$100 to ship a dozen or so to Australia. > > > > > > I'm willing to send you 2 transformers with each board blank ordered for > > $5.00US extra plus shipping. Please contact me off list for details. > > Sounds like an idea, I'll talk to you off list about it. > > > > > > > > Is it hard to support different FXS devices? I'm not sure exactly how > > > entwined app_rpt is with driver code. The reason I ask is one possible > > > avenue for me is some clone TDM cards from http://www.openvox.com.cn > > > but it looks like their TDM1200P clone uses a different driver. Their > > > 400P uses the standard digium driver. > > > > As long as the FXS device is seen as a Zaptel device in Asterisk, it will work > > with app_rpt. In our systems, we use ARIB cards with Digium TE110p/T100p T1 > > cards and a channel bank. It doesn't matter which driver is required as long > > is it is a Zaptel driver. > > > > Ah okay, that clears it up a little. I was planning on saving up for quad > radio card, especially for its DCS and CTCSS capabilities. However, I have > been offered a T1 card and channel bank for an unrefusable price, so I'm > probably going that way. > > > Regards, > > Jon > _______________________________________________ > App_rpt mailing list > App_rpt(a)lists.illiana.net > http://lists.illiana.net/mailman/listinfo/app_rpt >

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[App_rpt] Interface questions
by jon+apprpt＠unicomsystems.com 28 Jun '06

28 Jun '06

Hello all, First off, I must say that app_rpt looks like a very promising project and this style or repeater controller looks like the way of the future. I have a few questions that I couldn't quite come up with an answer for myself with a high degree of certainty, using the existing material online. 1). Does the 4 port radio interface card allow for 4 duplex repeater channels, or 2 duplex repeater channels? 2). The 600 ohm line transformer as used in the analogue radio interface appears to be out of stock on Digikey. Does anyone have alternative stockists or transformer models? Or even spares they'd be willing to sell? 3). Is app_rpt compatible with the TDM1200P or similar? I also have a few more developmental/roadmap oriented questions, namely, would it be possible for an interface to be developed that used a full duplex sound card and a serial or other data port to interface to a radio? Or could an interface be developed that had the necessary functions so that it only needed an FXO port on the Asterisk side, seeing FXOs can be got from even basic devices like some winmodems? Note that I'm not trying to rubbish the project or its developers with these questions, I'm sure the current interfaces work very well, but seeing as we're not interfacing phones, I thought it would've been possible design a way to interface at an even lower price point, opening access further, and further benefiting app_rpt with greater uptake levels from all sorts of radio people around the world. Best Regards, Jon

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[App_rpt] Autorun macro
by clayton.hull 26 Jun '06

26 Jun '06

Is there any way to run a macro in app_rpt automatically on startup? (For example, to command a node to link to other nodes automatically.) Clayton

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Re: [App_rpt] app_rpt.c test version 0-46-6
by hwstar＠rodgers.sdcoxmail.com 06 Jun '06

06 Jun '06

Eric, Glad to see you are up and runnign with 0.47. Will you be linking with RF to the remote site, or are you planning on doing the link with app_rpt over the Internet? Steve WA6ZFT > > From: Eric Tichansky <eric(a)lylix.net> > Date: 2006/06/06 Tue AM 07:18:32 PDT > To: Asterisk Repeater Controler <app_rpt(a)lists.illiana.net> > Subject: Re: [App_rpt] app_rpt.c test version 0-46-6 > > Super. Just built version 0.47 from SVN on my Asterisk 1.2.5 system. > The new autopatch features are excellent additions! Thumbs up! > > I finally acquired two mini HTs (2M and 440) from a recent local > hamfest, and have been able to do alot of testing. Between the > additional duplex states and the new autopatch features, the system is > quite robust and fits the bill perfectly. It is nice to be able to run > a full duplex autopatch w/o RX audio injection to the transmit side. > Great work. Eventually we plan to setup remote links to another app_rpt > enabled repeater for greater coverage. > > Eric NI3S > > Steve Rodgers wrote: > > Eric, > > > > Yes app_rpt version 0.46 is in Digium SVN as revision 29935 or later. > > Better to use the the released version as it has a notch filtering feature as > > well. > > > > Steve. > > > > > -- > > Eric Tichansky > Administrator > LYLIX.NET > http://lylix.net/ > > (724) 602-2264 > > _______________________________________________ > App_rpt mailing list > App_rpt(a)lists.illiana.net > http://lists.illiana.net/mailman/listinfo/app_rpt >

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[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 08 May '06

08 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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