- App_rpt-users - lists.keekles.org

talk faster
by Ken 18 Feb '15

18 Feb '15

yes - see http://ohnosec.org/drupal/node/142 for recording locations you can use Audacity to speed up the existing recordings (without changing the pitch of the voice) From: Skyler F <electricity440(a)gmail.com> To: app_rpt-users(a)ohnosec.org <app_rpt-users(a)ohnosec.org> Subject: [App_rpt-users] Faster telemetry Date: Wed, 18 Feb 2015 09:27:32 -0700 Is it possible for Allison to speak a bit faster when it's like "node xyz connected to node 40764"? Maybe wait less time in between the numbers when you connect or add my own audio files that are shorter? -- Skyler Fennell amsatnet.info KDØWHB electricity440(a)gmail.com _______________________________________________ App_rpt-users mailing list App_rpt-users(a)ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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Is it legal; Can it be done?
by Wayne 18 Feb '15

18 Feb '15

I have always wondered if it is legal to run a 2 meter –> HF net monitored and controlled by an appropriately licensed control operator. The 2 meter side is easy. I suspect the HF side is just as easy with VOX and a hysteresis based squelch circuit instead of the usual RF squelch. I know that controllers like a Arcom RC-210 is capable of basic HF rig control (suggesting legality?), but what about Allstar?

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AllStar Link SIP Trunk in FreePBX
by James Lyles 18 Feb '15

18 Feb '15

I have a raspberry pi running FreePBX/Asterisk with a VOIP mesh network. The internal extensions and Google voice trunk are working properly. I have an outgoing dial plan configured so that 70 (should go to interactive menu), 7xxxx or 7xxxxx (xxxx and xxxxx are node numbers) will dial the AllStar Link SIP trunk. The outgoing calls will ring the AllStar trunk and often, but not always, the call connects. If it connects, it usually does not have received audio. If there is audio, then when I enter node numbers, etc it appears the DTMF tones are not recognized. I say this because every node number I enter is "invalid". I can dial the AllStar access number, 763-230-0000, from the same phone over the Google voice trunk and everything works fine. I can also connect to the AllStar Link SIP directly using a software phone on my same LAN and it works fine. I'm assuming I am doing something wrong in the trunk configuration. Any help would be great. Thanks- James --------------------- I have it configured in FreePBX as follows: Trunk Name: AllStar Link PEER Details: username=1 fromuser=1 secret=allstar host=sip.allstarlink.org fromdomain=sip.allstarlink.org type=friend context=from-trunk insecure=port,invite trustrpid=yes directmedia=no keepalive=45 nat=yes dtmfmode=rfc2833 disallow=all allow=g711&ulaw USER Context: (no entry) USER Details: (no entry) Register String: 1:allstar@sip.allstarlink.org

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ACID VoIP Audio Bandwidth
by kk6ecm 17 Feb '15

17 Feb '15

How much audio bandwidth (highest useful frequency passed) can be expected using ACID via the VoIP link across the internet? What are the controlling/limiting factors? Thanks, Bob kk6ecm Sent from iPad

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app_rpt logging
by Pawlowski, Adam 17 Feb '15

17 Feb '15

Afternoon all, Is there a way to enable broad logging of connect, disconnect, keying, that's seen by app_rpt? Even cranked up to debug, the information about who's connected, say via echolink, is not available. I'd have to dump console output to a file, and I'm hoping I'm just missing some setting to get by this. Thoughts? Adam, KC2YQF

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Wireless links
by Robert Newberry 17 Feb '15

17 Feb '15

This may be viewed as an off topic, but I think it's worth talking about since it has to do with building our systems out. What is out there that is affordable for hams to connect our sites using microwave or some sort of non line of site technology? I would like to link our sites together when there is not a commercial internet connection available. Also this is a concern for keeping a network up and connected in the absence of commercial broadband due to natural or man made disaster.

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Re: [App_rpt-users] App_rpt logging
by Bryan Fields 17 Feb '15

17 Feb '15

On 2/17/15 4:11 PM, K Yoksh wrote: > > Have you tried my Swissarmy add-on for Asterisk? It will handle logging > among other features. > > www.qsl.net/k0kn/swissarmy <http://www.qsl.net/k0kn/swissarmy> > :neet: I've always wanted to try this out, never had the time. This looks like it's only run at connect and disconnect, so it shouldn't be an additional load on the box. 73's -- Bryan Fields 727-409-1194 - Voice 727-214-2508 - Fax http://bryanfields.net

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