- App_rpt-users - lists.keekles.org

Re: [App_rpt-users] Dial plan question
by Doug Crompton 16 Nov '13

16 Nov '13

Mike, Yes, the iax connection works and I can dial 3 digit extensions with the dial plan I showed. I just did not want to have to recreate all of the dial plans I have on the pbx server on the Allstar server. I wanted to wholesale send everything that went in the radio patch there for dial plan analysis. Since some extensions are 3 digits and others are more I was stymied as to how to do that. Possibly SIP would do a better job at this. Early dial is a sip feature I believe. It would not make much difference if I used IAX or SIP on the local circuit from a loading and routing standpoint. If app_rpt did not send the digits to the dial plan until the radio carrier detect was released this would work. The digits would not be processed until they were all present. Then a dial plan of _x! would work. It would just except it as sent and then dial the pbx server with the extension of whatever length it was. Perhaps there is a way to do this??? Maybe we need another option in autopatch to do this! 73 Doug WA3DSP http://www.crompton.com/hamradio From: mike(a)midnighteng.com To: doug(a)crompton.com Subject: RE: [App_rpt-users] Dial plan question Date: Fri, 15 Nov 2013 22:04:57 -0700 You must route the incoming digits to the other server. I used this method for my phone patch. I trunked it to my phone pbx (asterisk) for outgoing calls.You just need to make a iax connection to/from each system and check that it is functional with a route in your dialplan to the other server.As far as early dial ? -not sure if you can achieve what I think you mean by that without a bunch of script but you could start off with a prefix dial for route to the other server. That is how I make phone patches and give myself access to the radiopbx via phonepbx/inbound or sip. ...mike/kb8jnm -------- Original Message -------- Subject: Re: [App_rpt-users] Dial plan question From: Doug Crompton <doug(a)crompton.com> Date: Fri, November 15, 2013 11:30 pm To: "app_rpt-users(a)ohnosec.org" <app_rpt-users(a)ohnosec.org> Further info... What I want to achieve here is called "early dial" when connecting a sip phone to Asterisk. Each digit is sent to the server and if it does not match a dial plan it looks for the next digit until it finally finds a match. Is there a way to do this between servers to achieve what I asked in the prior message? 73 Doug WA3DSP http://www.crompton.com/hamradio From: doug(a)crompton.com To: app_rpt-users(a)ohnosec.org Date: Fri, 15 Nov 2013 22:04:33 -0500 Subject: [App_rpt-users] Dial plan question I have been using Asterisk for years but I just ran into a problem that I am sure there is a way to fix but I can't figure it out. I am connecting my Allstar server to my PBX asterisk server via IAX2. The connection works fine but I need some help on passing the extension. I want to pass the variable length extension to my pbx server. Extensions could be any combination of 3 digits up to a full phone number length. The actual extensions are processed in the PBX. Here is the Allstar extension snippet - [pbx_server] exten => _xxx,1,Answer exten => _xxx,n,noop("${EXTEN}") exten => _xxx,n,Wait(2) exten => _xxx,n,Dial(IAX2/pbx/${EXTEN}) exten => _xxx,n,Playback(vm-nobodyavail) exten => ,n,Hangup in rpt.conf 61=autopatchup,context=pbx_server,noct=1,farenddisconnect=1,dialtime=20000,quiet=1 ; Autopatch up 0=autopatchdn ; Autopatch down In the example I have put a 3 digit extension. So if I dial *61522 - 522 is passed to the pbx server. This works fine but I want to pass any length extension. If I put something like _x! - it just hits on the first digit. Maybe I could use waitexten() to capture it and pass to the pbx. Any ideas? The Allstar Asterisk and pbx Asterisk are obviously different machines on the same network. 73 Doug WA3DSP http://www.crompton.com/hamradio _______________________________________________ App_rpt-users mailing list App_rpt-users(a)ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users _______________________________________________ App_rpt-users mailing list App_rpt-users(a)ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

1 0

Dial plan question
by Doug Crompton 15 Nov '13

15 Nov '13

I have been using Asterisk for years but I just ran into a problem that I am sure there is a way to fix but I can't figure it out. I am connecting my Allstar server to my PBX asterisk server via IAX2. The connection works fine but I need some help on passing the extension. I want to pass the variable length extension to my pbx server. Extensions could be any combination of 3 digits up to a full phone number length. The actual extensions are processed in the PBX. Here is the Allstar extension snippet - [pbx_server] exten => _xxx,1,Answer exten => _xxx,n,noop("${EXTEN}") exten => _xxx,n,Wait(2) exten => _xxx,n,Dial(IAX2/pbx/${EXTEN}) exten => _xxx,n,Playback(vm-nobodyavail) exten => ,n,Hangup in rpt.conf 61=autopatchup,context=pbx_server,noct=1,farenddisconnect=1,dialtime=20000,quiet=1 ; Autopatch up 0=autopatchdn ; Autopatch down In the example I have put a 3 digit extension. So if I dial *61522 - 522 is passed to the pbx server. This works fine but I want to pass any length extension. If I put something like _x! - it just hits on the first digit. Maybe I could use waitexten() to capture it and pass to the pbx. Any ideas? The Allstar Asterisk and pbx Asterisk are obviously different machines on the same network. 73 Doug WA3DSP http://www.crompton.com/hamradio

1 1

Re: [App_rpt-users] No TX audio upon reboot after initial install. Slow DMK URI LED Blink (Detailed info)
by William Schleppegrell 14 Nov '13

14 Nov '13

Will attempt to set processor for maximum performance in BIOS tonight and see if it helps. Sent from my Windows Phone From: Bryan D. Boyle Sent: 11/14/2013 11:33 AM To: David Powell-Williams Cc: <app_rpt-users(a)ohnosec.org> Subject: Re: [App_rpt-users] No TX audio upon reboot after initial install. Slow DMK URI LED Blink (Detailed info) of course, if there was less talking on the repeater, one would have to ask why have a repeater up at all? then you need NO horsepower to run something that doesn't exist. -- Bryan Sent from my iPhone 5...small keyboard, big fingers...please forgive misspellings... On Nov 14, 2013, at 12:18, "David Powell-Williams" <dpowellwilliams(a)shaw.ca> wrote: > Of course if there was less talk on the repeater, the computer would require > less hoarse-power :) > > > _______________________________________________ > App_rpt-users mailing list > App_rpt-users(a)ohnosec.org > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users _______________________________________________ App_rpt-users mailing list App_rpt-users(a)ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

1 0

Re: [App_rpt-users] No TX audio upon reboot after initial install. Slow DMK URI LED Blink (Detailed info)
by Bryan D. Boyle 14 Nov '13

14 Nov '13

of course, if there was less talking on the repeater, one would have to ask why have a repeater up at all? then you need NO horsepower to run something that doesn't exist. -- Bryan Sent from my iPhone 5...small keyboard, big fingers...please forgive misspellings... On Nov 14, 2013, at 12:18, "David Powell-Williams" <dpowellwilliams(a)shaw.ca> wrote: > Of course if there was less talk on the repeater, the computer would require > less hoarse-power :) > > > _______________________________________________ > App_rpt-users mailing list > App_rpt-users(a)ohnosec.org > http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

1 0

Re: [App_rpt-users] No TX audio upon reboot after initial install. Slow DMK URI LED Blink (Detailed info)
by David Powell-Williams 14 Nov '13

14 Nov '13

Of course if there was less talk on the repeater, the computer would require less hoarse-power :)

1 0

No TX audio upon reboot after initial install. Slow DMK URI LED Blink (Detailed info)
by William Schleppegrell 14 Nov '13

14 Nov '13

My node consists of an Intel NUC DCCP847DYE PC Which has an Intel Celeron™ Processor 847(1.10 GHz, 2M Cache) It has 3 USB ports and Video / Audio out are supported by 2 HDMI ports which may be part of my problem. • Intel® HD Graphics • Dual HDMI Ports supporting dual independent display Intel® High Definition Audio (Intel® HD Audio) via two HDMI 1.4a outputs supporting 8 channel (7.1) digital audio I am running a single node on this pc. The node is operating a UHF repeater. Two radios. Icom Commercial IC-F420 mobiles via a DMK URI. I loaded ACID and my system ran fine until I rebooted. This is my second attempt at installation. I will include my .conf files below. I have PL tones set on radios but am not using COR/COS. I utilize the software to detect PL to trigger TX/RX. I have a slow blinking green LED on the DMK URI 2-3 flashes per second, when it has failed and is not sending audio to the TX radio(like now). And a nice fast blink pattern when it is working. Ok...Oh the is keying up with a staticy transmission during ID's and when Flash is run in the radio-tune-menu, but not recognizable audio is heard. RPT.CONF [29922] ; Change this to your assigned node number rxchannel=Radio/usb29922 duplex=2 erxgain=-3 etxgain=3 ;controlstates=controlstates scheduler=schedule29922 morse=morse29922 macro=macro29922 functions=functions29922 phone_functions=functions29922 link_functions=functions29922 telemetry=telemetry wait_times=wait-times context = radio callerid = "Repeater" <0000000000> idrecording = |iAE5YY accountcode=RADIO hangtime=100 althangtime=100 totime=170000 idtime=540000 politeid=3000 idtalkover=|iAE5YY unlinkedct=ct2 remotect=ct3 linkunkeyct=ct8 ;nolocallinkct=0 ;eannmode=1 ;connpgm=yourconnectprogram ;discpgm=yourdisconnectprogram ;lnkactenable=0 ;lnkacttime=1800 ;lnkactmacro=*52 ;lnkacttimerwarn=30seconds ;remote_inact_timeout=1800 ;remote_timeout=3600 nounkeyct=0 ;holdofftelem=0 beaconing=0 ; ; *** Status Reporting *** ; ; Uncomment the either group following two statpost lines to report the status of your node to stats.allstarlink.org ; depending on whether you are running ACID or Limey Linux. ; ** For ACID *** statpost_program=/usr/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates ; ** For Limey Linux ** ;statpost_program=/bin/wget,-q,--timeout=15,--tries=1,--output-document=/dev/null ;statpost_url=http://stats.allstarlink.org/uhandler.php ; Status updates ; ; Morse code parameters, these are common to all repeaters. ; [morse29922] speed=18 frequency=800 amplitude=4096 idfrequency=750 idamplitude=1024 [schedule29922] ;dtmf_function = m h dom mon dow ; ala cron, star is implied [functions29922] 1=ilink,1 2=ilink,2 3=ilink,3 4=ilink,4 5=macro,1 70=ilink,5 71=ilink,6 72=ilink,7 73=ilink,15 74=ilink,16 75=ilink,8 80=status,1 81=status,2 6=autopatchup,noct=1,farenddisconnect=1,dialtime=20000 ; Autopatch up 0=autopatchdn ; Autopatch down ;989=cop,4 980=status,3 99=cop,6 ; Place command macros here [macro29922] [telemetry] ct1=|t(350,0,100,2048)(500,0,100,2048)(660,0,100,2048) ct2=|t(660,880,150,2048) ct3=|t(440,0,150,4096) ct4=|t(550,0,150,2048) ct5=|t(660,0,150,2048) ct6=|t(880,0,150,2048) ct7=|t(660,440,150,2048) ct8=|t(700,1100,150,2048) remotetx=|t(1633,0,50,3000)(0,0,80,0)(1209,0,50,3000); remotemon=|t(1209,0,50,2048) cmdmode=|t(900,903,200,2048) functcomplete=|t(1000,0,100,2048)(0,0,100,0)(1000,0,100,2048) patchup=rpt/callproceeding [wait-times] telemwait=2000 idwait=500 unkeywait=2000 calltermwait=2000 ; ; This is where you define your nodes which cam be connected to. ; [nodes] ; Note, if you are using automatic update for allstar link nodes, ; no allstar link nodes should be defined here. Only place a definition ; for your locak nodes, and private (off of allstar link) nodes here. 29922 = radio(a)127.0.0.1/29922,NONE ; Memories for remote bases [memory] ;00 = 146.580,100.0,m ;01 = 147.030,103.5,m+t ;02 = 147.240,103.5,m+t ;03 = 147.765,79.7,m-t ;04 = 146.460,100.0,m ;05 = 146.550,100.0,m USBRADIO.CONF [general] [usb29922] hdwtype=0 rxboost=1 txboost=1 rxctcssrelax=1 txctcssdefault=136.5 rxctcssfreqs=136.5 txctcssfreqs=136.5 ;rxctcssoverride=0 carrierfrom=dsp ctcssfrom=dsp rxdemod=flat txprelim=no txlimonly=no txtoctype=no txmixa=composite txmixb=no invertptt=0 duplex=1 rxondelay=0 rxnoisefiltype=0 eeprom=0 #includeifexists custom/usbradio.conf IAX.CONF ; Inter-Asterisk eXchange driver definition ; ; [general] bindport=4569 bindaddr=0.0.0.0 disallow=all allow=gsm allow=g726aal2 allow=ulaw jitterbuffer=yes forcejitterbuffer=yes dropcount=2 maxjitterbuffer=4000 maxjitterinterps=10 resyncthreshold=1000 maxexcessbuffer=80 minexcessbuffer=10 jittershrinkrate=1 tos=0x1E autokill=yes delayreject=yes ;iaxthreadcount=30 ;iaxmaxthreadcount=150 register=29922:<PASSWORD OMITTED>@register.allstarlink.org IN EMAIL for SECURITY [radio-proxy] type=user deny=0.0.0.0/0.0.0.0 permit=/255.255.255.255 context=radio-secure-proxy disallow=all allow=g726aal2 transfer=no [radio-proxy-out] type=peer host= username=29922 secret=<OMITTED for SECURITY> auth=md5 disallow=all allow=g726aal2 transfer=no ; Incoming radio connections [radio] type=user disallow=all allow=g726aal2 allow=gsm codecpriority=host context=radio-secure transfer=no ; Incoming phone connections to app_rpt ;[radio-control] ;type=user ;secret=nunya ;disallow=all ;allow=gsm ;codecpriority=host ;context=radio-control ;transfer=no ; Setup for outgoing calls ;[yourprovider] ; This needs to change ;type=peer ;host=127.0.0.1 ;secret=nunya ;auth=md5 ;disallow=all ;allow=gsm ;transfer=no [allstar-sys] type=user context=allstar-sys auth=rsa inkeys=allstar disallow=all allow=ulaw [allstar-public] type=user context=allstar-public auth=md5 secret=allstar disallow=all allow=ulaw allow=gsm EXTENSIONS.CONF [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here. [globals] HOMENPA=713 [default] exten => i,1,Hangup [radio-secure] exten => 29922,1,rpt,29922 [radio-secure-proxy] exten => 29922,1,rpt,29922|X exten => _0X.,1,Goto(allstar-sys|${EXTEN:1}|1) [pstn-out] exten=_NXXNXXXXXX,1,playback(ss-noservice) exten=_NXXNXXXXXX,2,Congestion [invalidnum] exten=s,1,Wait,3 exten=s,n,Playback,ss-noservice exten=s,n,Wait,1 exten=s,n,Hangup [radio] exten=_X11,1,Goto(check_route|${EXTEN}|1); exten=_NXXXXXX,1,Goto(check_route|1${HOMENPA}${EXTEN}|1) exten=_1XXXXXXXXXX,1,Goto(check_route|${EXTEN}|1) exten=_07XX,1,Goto(parkedcalls|${EXTEN:1}|1) exten=00,1,Goto(my-ip|s|1) [check_route] ; no 800 exten=_1800NXXXXXX,2,Goto(invalidnum|s|1) exten=_1888NXXXXXX,2,Goto(invalidnum|s|1) exten=_1877NXXXXXX,2,Goto(invalidnum|s|1) exten=_1866NXXXXXX,2,Goto(invalidnum|s|1) exten=_1855NXXXXXX,2,Goto(invalidnum|s|1) ; no X00 NPA exten=_1X00XXXXXXX,2,Goto(invalidnum|s|1) ; no X11 NPA exten=_1X11XXXXXXX,2,Goto(invalidnum|s|1) ; no X11 exten=_X11,2,Goto(invalidnum|s|1) ; no 555 Prefix in any NPA exten=_1NXX555XXXX,2,Goto(invalidnum|s|1) ; no 976 Prefix in any NPA exten=_1NXX976XXXX,2,Goto(invalidnum|s|1) ; no NPA=809 exten=_1809XXXXXXX,2,Goto(invalidnum|s|1) ; no NPA=900 exten=_1900XXXXXXX,2,Goto(invalidnum|s|1) ; okay, route it exten=_1NXXXXXXXXX,1,Goto(pstn-out|${EXTEN:1}|1) exten=_X.,2,Goto(invalidnum|s|1) [my-ip] exten=s,1,Set(MYADDR=${CURL(http://myip.vg)}) exten=s,2,Wait,1 exten=s,3,SayAlpha(${MYADDR}) exten=s,4,Hangup [allstar-sys] exten => 0,1,Answer exten => 0,n,System(rm -rf /tmp/allstar-config\; mkdir -p /tmp/allstar-config\;curl -sk -m 20 --retry 1 https://config.allstarlink.org/config/start.sh > /tmp/allstar-config/start.sh\;sh /tmp/allstar-config/start.sh ${CALLERID(name)}) exten => 0,n,Hangup exten => _0.,1,Answer exten => _0.,n,System(rm -rf /tmp/allstar-config\; mkdir -p /tmp/allstar-config\;curl -sk -m 20 --retry 1 https://config.allstarlink.org/config/start.sh > /tmp/allstar-config/start.sh\;sh /tmp/allstar-config/start.sh ${CALLERID(name)}) exten => _0.,n,Hangup exten => _1.,1,Rpt(${EXTEN:1}|Rrpt/node:NODE:rpt/in-call:digits/0:PARKED|120) exten => _1.,n,Hangup exten => _2.,1,Ringing exten => _2.,n,Wait(3) exten => _2.,n,Answer exten => _2.,n,Playback(rpt/node) exten => _2.,n,Saydigits(${EXTEN:1}) exten => _2.,n,Rpt(${EXTEN:1}|P|${CALLERID(name)}-P) exten => _2.,n,Hangup exten => _3.,1,Ringing exten => _3.,n,Wait(3) exten => _3.,n,Answer exten => _3.,n,Playback(rpt/node) exten => _3.,n,Saydigits(${EXTEN:1}) exten => _3.,n,Rpt(${EXTEN:1}|Pv|${CALLERID(name)}-P) exten => _3.,n,Hangup exten => _4.,1,Ringing exten => _4.,n,Wait(3) exten => _4.,n,Answer exten => _4.,n,Playback(rpt/node) exten => _4.,n,Saydigits(${EXTEN:1}) exten => _4.,n,Rpt(${EXTEN:1}|D|${CALLERID(name)}-P) exten => _4.,n,Hangup exten => _5.,1,Ringing exten => _5.,n,Wait(3) exten => _5.,n,Answer exten => _5.,n,Playback(rpt/node) exten => _5.,n,Saydigits(${EXTEN:1}) exten => _5.,n,Rpt(${EXTEN:1}|Dv|${CALLERID(name)}-P) exten => _5.,n,Hangup [allstar-public] exten => s,1,Ringing exten => s,n,Set(RESP=${CURL( https://register.allstarlink.org/cgi-bin/authwebphone.pl?${CALLERID(name)}) }) exten => s,n,Set(NODENUM=${CALLERID(number)}) exten => s,n,GotoIf($["${RESP:0:1}" = "?"]?hangit) exten => s,n,GotoIf($["${RESP:0:1}" = ""]?hangit) exten => s,n,GotoIf($["${RESP:0:5}" != "OHYES"]?hangit) exten => s,n,Set(CALLSIGN=${RESP:5}) exten => s,n,Wait(3) exten => s,n,Playback(rpt/node|noanswer) exten => s,n,Saydigits(${NODENUM}) exten => s,n,Set(CALLERID(num)=0) exten => s,n,Set(CALLERID(name)=${CALLSIGN}) exten => s,n,Rpt(${NODENUM}|X) exten => s,n,Hangup exten => s,n(hangit),Answer exten => s,n(hangit),Wait(1) exten => s,n(hangit),Hangup exten => 9999,1,Goto(allstar-public|s|1) USBRADIO_TUNE_USB29922.CONF [usb29922] ; name=usb29922 ; devicenum=1 devstr=2-1.1 rxmixerset=500 txmixaset=300 txmixbset=500 rxvoiceadj=0.500000 rxctcssadj=0.500000 txctcssadj=200 rxsquelchadj=799 Please feel free to contact me (William) at any time on my cell @ 832-FIVE 4 4-9810

3 2

Advise for setup
by Angelo Glorioso 14 Nov '13

14 Nov '13

Good All, We have a node coming up in Lake Charles, La and we are having issues. So, I am asking for configurations suggestion. First off, the Allstar node is not going to be at the repeater site. It is at a remote site. It is running on GM300 with URI interface. The repeater has a long squelch tail of about 2 second. I have tried to run with USB radio in half duplex mode but the node ends up ping ponging with the repeater in half duplex mode . Is it better to use simpleusb or USBradio for this type of setup?? Should the repeater squelch be reduced or turned off all together. Please offer a copy of either USBradio or simpleusb configuratio file pretty please. Any help would be appreciated. 7 3 de Angelo

5 6

RTCM, how do I use it in Simplex?
by Shaun Rudling 12 Nov '13

12 Nov '13

Please open my voter.conf I have 4 RTCM units live at the moment running as full-duplex repeaters and the AllStar system/RTCM combo works very well. But I want to add a 5'th RTCM but this one is in simplex mode. It must either RX or TX, not both. It is a single 6m transceiver. It must not TX while it is receiving. But as soon as it starts receiving it will key up. Somehow it must be told not to TX while in RX. The new simplex unit is shown as "8G6M = 8G6M,transmit" in the attached voter.conf file If it is not possible, I might have to purchase another 6m transceiver and just have it as a 6m repeater. Shaun

2 1

beaglebone w/ URI
by Greg 12 Nov '13

12 Nov '13

Any possibility Pickle or Limey would run on the beaglebone black with usb URI? Will that board power the URI? Just an idea..

3 2

Modified USB Audio Problems on Acid install
by Richard Ainley 12 Nov '13

12 Nov '13

Hello, Im very new to this and although I have spent the past week or so trying to fix these issues myself I have to now admit im stuck! Im still learning when it comes to linux!! The problems I have are as follows, im not sure who to ask for help hence my general request here! * Having installed a modified USB dongle I am unable to get any receive sound (usb microphone input) to appear in the web transceiver application. I have run the radio tune rxnoise, rxvoice and see the stats changing but still cannot get radio rx sound over the internet. Audio from the web transceiver to my radio works just great, along with the PTT. * Im also having problems connecting to my node via the SIP portal, I have a working SIP Telephone setup which I have tested to other nodes so I know the SIP hardware is ok however, when I connect to my node 27308 I get the usual welcome message, am asked to enter my pin etc and select 2 for manual operation the call goes silent and then eventually disconnects. I have forwarded port 4569 to my server IP. * Lastly the UK dial in telephone number appears to be not working with the message node not available, no welcome message just straight to that message!! Thanks very much for reading my email having exhausted google searches in circles I'm hoping you guys will point out my hopefully silly mistakes!! 73 Rich G7WIZ - Grimsby UK

2 1