- App_rpt-users - lists.keekles.org

New version of NodeShark available with remote script execution feature
by Keith Williamson 04 Dec '10

04 Dec '10

Hello All, I have uploaded a greatly improved version (v0.95) of NodeShark. This version adds a new remote shell script execution feature allowing you to remotely execute any non-interactive shell script on your AllStar system with a single click. The shell script output is displayed in a command monitor window. The new feature includes a simple shell script editor that allows you to create, name, view, edit, and delete custom scripts from within NodeShark. You can create as many scripts as you like and they are added to a convenient drop-down list for single click execution. You can also import shell scripts created outside of NodeShark using other editors. Finally, v0.95 is the result of extensive code cleanup and highly improved error checking. I have also updated the NodeShark User Guide to reflect the new features. You can download this latest version and the user guide at http://www.arizuni.com/NodeShark Currently, NodeShark directly supports multi-node to only a single AllStar system. The final feature I'll be adding prior to 1.0 is AllStar multi-site support (i.e multiple selectable configurations). 73's, Keith KF7DRV AllStar: 2541 Echolink: 477448

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Web Transceiver Telemetry
by Greg Thornwall 02 Dec '10

02 Dec '10

I wanted to turn off announcements for the web transceiver I setup my rpt.conf file as follows: telemetry=telemetry ... [telemetry] eannmode=0 ; echolink telemetry guilinkdefault=0 ; gui telemetry telemdefault=0 ; local telemetry phonelinkdefault=0 ; phone telemetry But I still hear the telemetry. Either I have the parameters wrong or is that controlled by the Web Transceiver? Also, is there a way to automatically connect to a node once asterisk is restarted or after a reboot? Thanks, Greg

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PICO NODE is now a REALITY!
by Jim Duuuude 02 Dec '10

02 Dec '10

We (Dale Weber, N7PKT and I) have been working on porting Asterisk/app_rpt to the TINY (about 3.5 x 3.5 inches) BeagleBoard (arm-based Linux system), using the native (built-in) audio and GPIO hardware to directly interface to radio systems. As of this evening, we have success!! The BeagleBoard is capable of talking to 2 radios (either half or full duplex) using the radio's COR and CTCSS decoders (sorry, there isnt enough CPU power to do the DSP processing, like in chan_usbradio). It will also talk to a URI (or 2 or maybe even 3??) using chan_simpleusb. I am designing an I/O card, which properly conditions both the analog and digital signals in and out of the BeagleBoard, and will provide general-purpose GPIO pins for whatever the user's needs are. It also may contain a DC-DC converter, so that the system may be powered from a 12 volt (well, like maybe 9-18 volt, actually) DC input, allowing use in an automobile and various configuration of battery/solar/ wind-power. For those interested, I posted the working chan_beagle.c source in the SVN, and will update it as the project moves along. Currently it only supports 1 fully operational channel, but that's because my test hardware was not intended to do anything other then that. We will also post the working kernel sources, plus an entire image of the 4GB flash card, for ease of installation. JIM WB6NIL

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60 DAY WARNING
by Jim Duuuude 01 Dec '10

01 Dec '10

WARNING -- All UN-MIGRATED NODES WILL BE PURGED OUT OF THE DATABASE AFTER 1/31/11!!!! PLEASE be sure to migrate your nodes now. Dont wait until the last minute!!! Thanks JIM WB6NIL

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Third Party Call Control through AMI
by Max Godoy 30 Nov '10

30 Nov '10

Hi everybody! We are trying to integrate radios in an Emergecy Call Center environement. We control the queues, users and vectoring of the system externally through the AMI API. We want to be able to mix phone calls and radio calls. The URI seems a good and inexpensive solution for small and medium setups (2 radios, a dozen phone lines and 4 agents). The problem is that we need to control the system from a central point using a Third Party Call Control strategy. We have chosen the AMI interface in Asterisk as it is aimed at this sort of operations. Has anybody interfaced the app_rpt module with AMI? Can anybody give us a lead or a hint on how to do that? Is there a different radio control system that interacts with AMI? Regards, Max *********************************************************** Max Godoy de la Barra CENTRO ESPAÑOL DE SERVICIOS TELEMATICOS, S.A. (CESTEL) c/ La Granja nº5 28500 Arganda del Rey (Madrid) Tfno: +34 91 871 20 95 ext. 220 Fax: +34 91 870 28 18 Email: mgodoy(a)cestel.es Web Site: http://www.cestel.es

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Phone System
by Greg Thornwall 29 Nov '10

29 Nov '10

How hard would it be to get allstar asterisk to work on our home phones? Voice mail, a phone patch to allstar maybe, on hold music etc. I guess we would need a PCI phone board to interface with the phones in the house and outside line (POT)? Any good detailed documentation on how to do this? tnx, 73, Greg

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Re: [App_rpt-users] No RX Audio on iaxRPT
by David Osborn 29 Nov '10

29 Nov '10

Rod, There's something lurking at the back of my mind (where there's a lot of dust!) which tells me that this is a behavioural thing between iaxRPT and a SIMPLEX node. Have you tried connecting to and working through another node? I had a locally connected repeater configured in simplex mode (I got it into my head to use internal talkthrough, bypassing the DSP). When I connected using iaxRPT I got exactly the same symptoms as you have reported. Now that the node has been configured to full-duplex, iaxRPT works just fine. You may not have done anything wrong after all! Dave GD4HOZ (2480 - 2484)

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No RX Audio on iaxRPT
by Rod Schnell 28 Nov '10

28 Nov '10

I'm sure this problem arises frequently (and I have read and tried most solutions in this and other places). SETUP: Motorola M1225 w/ flat audio in simplex mode -->> URI adapter -->> Asterisk (CentOS ISO from AllStarLink link) I have no RX audio on iaxRPT. I have tried different radios & different computers. Here's what is happening: 1) start asterisk and go in to the CLI at verbose level 3 or 4 (at this point the node appears on the AllstarLink active list) 2) start iaxRPT, with debug level at 2 3) connect to node via iaxRPT. get the following * Verbosity is at least 3 -- Hungup 'IAX2/gui-4704' -- Hungup 'Zap/pseudo-1686370267' -- Accepting AUTHENTICATED call from 68.95.77.100: > requested format = gsm, > requested prefs = (), > actual format = gsm, > host prefs = (ulaw|gsm), > priority = mine -- Executing [27234@radio-gui:1] Rpt("IAX2/gui-3545", "27234|X") in new stack == Spawn extension (radio-gui, 27234, 1) exited KEEPALIVE on 'IAX2/gui-3545'* 4) click TRANSMIT on iaxRPT (or via radio), unkey. Get tone then ID over both radio and iaxRPT. 5) TX on handheld radio and do test count. NO audio heard on iaxRPT. Audio is heard on radio connected to URI adapter. When handheld unkeys I get the tone signaling end-of-transmission. (This leads me to understand that asterisk IS seeing the radio receive.) 6) with two computers running iaxRPT, when one TX the other one can hear it. (This leads me to believe that asterisk IS working.) So with all of the above I can only see two possibilities for things to be wrong, either the URI adapter has gone bad or I've not got something correct in the program. Any help would be greatly appreciated. usbradio.conf --original file comments removed so I could add what I've tested and results ; ; Usbradio channel driver Configuration File ; [general] [usb] eeprom=0 ; added to remove error in asterisk startup hdwtype=0 rxboost=1 rxctcssrelax=1 ;txctcssdefault= 100.0 ;rxctcssfreqs= 100.0 ;txctcssfreqs= 100.0 ;rxctcssoverride=0 ;rxctcssoverride=1 ;this added as test carrierfrom=dsp ctcssfrom=rxdemod ;tried this as 'no' and i get a loop of tx/unkey, repeat ;tried this as 'usb' go no change. last try was rxdemod before sending email rxdemod=flat txprelim=yes txlimonly=yes txtoctype=notone txmixa=voice txmixb=no ; have tried this as voice and auxvoice but no difference found invertptt=0 duplex=0 rxondelay=20

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Upgrading Asterisk / app_rpt
by Ramesh Dhami (VA3UV) 27 Nov '10

27 Nov '10

Hi All; Just tried to update one of my nodes by pulling the latest package off the SVN repository, using the following: wget http://x.allstarlink.org/svn-trunk.tar.gz -O- | tar xzv I get a 404 error. Perhaps the package is being updated??? Thanks, Ramesh.

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allstar hub
by Robert Guenther 26 Nov '10

26 Nov '10

Just created a server in my data center for use as a hub only. I was told to change rxchannel = zap/pseudo in rpt.conf, however when I go to the web transceiver and connect I hear the tones and then it sounds like packet loss and I get disconnected. Looking at my log file in asterisk I see the following. allstar*CLI> -- Accepting AUTHENTICATED call from x.x.x.x: > requested format = ulaw, > requested prefs = (ulaw), > actual format = ulaw, > host prefs = (ulaw|gsm), > priority = mine -- Executing [s@allstar-public:1] Ringing("IAX2/allstar-public-2179", "") in new stack -- Executing [s@allstar-public:2] Set("IAX2/allstar-public-2179", "RESP=OHYESW9RCG") in new stack -- Executing [s@allstar-public:3] Set("IAX2/allstar-public-2179", "NODENUM=27489") in new stack -- Executing [s@allstar-public:4] GotoIf("IAX2/allstar-public-2179", "0?hangit") in new stack -- Executing [s@allstar-public:5] GotoIf("IAX2/allstar-public-2179", "0?hangit") in new stack -- Executing [s@allstar-public:6] GotoIf("IAX2/allstar-public-2179", "0?hangit") in new stack -- Executing [s@allstar-public:7] Set("IAX2/allstar-public-2179", "CALLSIGN=W9RCG") in new stack -- Executing [s@allstar-public:8] Wait("IAX2/allstar-public-2179", "3") in new stack -- Executing [s@allstar-public:9] Playback("IAX2/allstar-public-2179", "rpt/node|noanswer") in new stack -- <IAX2/allstar-public-2179> Playing 'rpt/node' (language 'en') -- Executing [s@allstar-public:10] SayDigits("IAX2/allstar-public-2179", "27489") in new stack -- <IAX2/allstar-public-2179> Playing 'digits/2' (language 'en') -- <IAX2/allstar-public-2179> Playing 'digits/7' (language 'en') -- <IAX2/allstar-public-2179> Playing 'digits/4' (language 'en') -- <IAX2/allstar-public-2179> Playing 'digits/8' (language 'en') -- <IAX2/allstar-public-2179> Playing 'digits/9' (language 'en') -- Executing [s@allstar-public:11] Set("IAX2/allstar-public-2179", "CALLERID(num)=0") in new stack -- Executing [s@allstar-public:12] Set("IAX2/allstar-public-2179", "CALLERID(name)=W9RCG") in new stack -- Executing [s@allstar-public:13] Rpt("IAX2/allstar-public-2179", "27489|X") in new stack == Spawn extension (allstar-public, s, 13) exited KEEPALIVE on 'IAX2/allstar-public-2179' -- <Zap/pseudo-1579548112> Playing 'rpt/node' (language 'en') -- <Zap/pseudo-1579548112> Playing 'letters/w' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/9' (language 'en') -- <Zap/pseudo-1579548112> Playing 'letters/r' (language 'en') -- <Zap/pseudo-1579548112> Playing 'letters/c' (language 'en') -- <Zap/pseudo-1579548112> Playing 'letters/g' (language 'en') -- <Zap/pseudo-1579548112> Playing 'rpt/connected' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/2' (language 'en') -- <Zap/pseudo-1579548112> Playing 'rpt/node' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/2' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/7' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/4' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/8' (language 'en') -- <Zap/pseudo-1579548112> Playing 'digits/9' (language 'en') -- Hungup 'Zap/pseudo-1579548112' -- Hungup 'IAX2/allstar-public-2179' -- Hungup 'Zap/pseudo-273640313' -- <Zap/pseudo-692230668> Playing 'rpt/node' (language 'en') -- <Zap/pseudo-692230668> Playing 'letters/w' (language 'en') allstar*CLI> Why does Zap/pseudo hang up? After it hangs up on the web transceiver it sounds like a loop of non routable sounds. Thanks Robert W9RCG

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