June 2012 - App_rpt-users - lists.keekles.org

Re: [App_rpt-users] Dialing from RF to SIP/IAX CLIENT..
by Bradley Haney 27 Jun '12

27 Jun '12

Would anyone know how to intergate this if you already use the allstarlink autopatch? OUr club is using the alltarlink autopatch for 5.00 a month and we would like to add a few SIP/IAX extensions so the Rf users coudl dial out on autopatch and then also dial the extension numbers. Is it possible to have both or would i have to get rid of the allstarlink autopatch? Or is there an easy way to configure it so if you wanted to dial out on the autopatch you have to do a *6 for autopatch and then lets say 9 for an outside line? Bradley On Mon, Jun 11, 2012 at 1:41 PM, Monty Dana <ke7jvx(a)gmail.com> wrote: > This is what I used from Keith KF7DRV: > > http://ohnosec.org/pipermail/app_rpt-users/2010-February/000787.html > > Monty > > > On Sat, Jun 9, 2012 at 9:11 PM, Bradley Haney <kc9gqr(a)gmail.com> wrote: >> >> Hello all.. looked up the great info on getting a iax client working >> on our repeater.. I can dial our node number now from a IAX client on >> my phone and use *99 to tx and # to stop Tx and listen.. Now I >> would like to be able to "Call" my phone from the repeater. We are >> currently using allstarlink's Autopatch also.. so would i just dial >> *6 and then the extension number i created? >> or would i have to do more work in the extensions.conf dept. >> >> Thanks for all the help in advance.. >> >> Bradley >> _______________________________________________ >> 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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DMK uri FT8900 help
by ea8ee 27 Jun '12

27 Jun '12

Any info about the wiring to minidin 6pind yaesu ?

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Repeater Audio?
by Alan Adamson 27 Jun '12

27 Jun '12

isn't exactly Allstar related, but sorta. I've got a challenge that I'm not sure how to resolve. I think I've said this prior. I've successfully glued an RTCM to a New Icom NXDN repeater. This repeater is running in mixed analog mode and the Allstar is gated to analog only. There is no external controller running other than the RTCM. The internal controller functions when there is a digital signal and that doesn't come across allstar. For now that's an issue I'm willing to accept on that repeater. By using the Disc Out and the Mod in, this setup sounds *awesome*, I'm just blown away at how good it all sounds end to end. I'm about to head down this path on a second repeater, but this one I want to bring up as either Digital only or mixed digital. The controller has the above mentioned Disc out, but that only works with an analog signal. To get *transcoded * audio out, you have to use a pin called AF OUT, it's fixed level, but it's not discriminator, at best it sounds like speaker level. This is the only way to get an intelligent audio signal to feed to allstar when running in digital mode, there is no equivalent of discriminator out when in digital mode. Because I'm using an RTCM, I also can't *process the audio* or at least it's not obvious. And putting that audio back out into the network, it sounds really bad. it's too bassy, and muddy, basically it's de-emphasized and it really needs to be pre-emphasized. Is there any way with an RTCM to tell anything that I've got de-emphasized audio and to help it out? (it really needs some form of low pass put on it and the mid/high frequencies boosted). Also for note, I'm running my RTCM in conventional mode, not voter mode. and I have a CTCSS gate that is valid with COR *AND* CTCSS (in digital in NXDN they call that RAN codes). So I either need to filter between the radio and the RTCM or in the Allstar code if possible. BTW, I really have the same problem coming the other way, FM audio coming from Allstar and being then digitized to NXDN digital (AMBE+2 for those that want to know) *can* sound *really* good, but can also sound really crappy. If the person on the talking end is mumbling, or has a very deep voice, it doesn't sound worth crap, if on the other hand the person has a bright voice and talks clearly, it sounds amazing. This has nothing to do with the *emphasis* of this audio, it's FLAT going into the discriminator, the radio from there handles the transcoding. So, as a start, I'm looking for a relatively easy (and cheap) way to take speaker quality audio from the repeater and put it out *improved* on allstar with a RTCM. Note, with a URI, I could do that, but I don't seem to know how if possible with the RTCM/Voter/Rpt.conf stuff. Anyone faced a similar challenge and found an acceptable approach? I thought about trying one of the CAT audio boards, or maybe building a 3 or 5 band graphic equalizer, but I can't be the only person who's had this issue and solved it? Thanks in advance, and let me know your thoughts. ps. I'm happy to give anyone a demo, just let me know and I'll reconfigure my repeater for digital and show you. I'm testing some analog stuff at the moment, and the new repeater is setting on my floor currently. Alan

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RTCM Question
by Alan Adamson 26 Jun '12

26 Jun '12

Anyone else noticed this? Maybe my user radio is just a fast switcher. I use and RTCM in conventional mode at my repeater to be the interface for it and Allstar. If I don't remember to keep the PTT held for about a second after I finish talking, as soon as I release the PTT, I get about 1 second (it's actually a smidge shorter than that) of the last few things I said on the air. I'm pretty sure this *repeat* is because of the approx 1 second delay that the RTCM instills in the system. It's about 400ms of startup delay, and about the same of system delays, so the live release of the PTT sees approx 800ms of delayed audio. If I remember to keep the PTT pressed for approx 1 second when I finish talking I don't see this. I'm using a commercial radio, so maybe I'll need to see about a programmed setting to do that for me automagically. Just wondering if others are seeing it as well or if I just have a fast switcher. Alan

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Re: [App_rpt-users] RTCM Question
by Alan Adamson 26 Jun '12

26 Jun '12

Thanks Ken, I suspect that with the RTCM and the 2 buffers that need to be filled, I'm not sure this can be resolved so easily. It's talked about in the manual, and you wouldn't think you wouldn't notice a second of delay (finger being that quick to unkey - now key, sheesh, I think there is lightning responsiveness there :)), but it bites me every time. it's just become my "oh the allstar is functioning" detector, then I get the CT :). Alan From: Ken [mailto:ke2n@cs.com] Sent: Tuesday, June 26, 2012 8:26 AM To: 'Alan Adamson'; app_rpt-users(a)ohnosec.org Subject: RE: [App_rpt-users] RTCM Question I have seen this kind of thing between servers at the same location - without an RTCM - when something was causing some latency issues on my LAN. in iax.conf I found the default resyncthreshold= was 1000 ms so I reduced it. I have since fixed the problem with the LAN, but still run resyncthreshold at 300 ms to avoid ever building up a lot of delay in the system. 73 Ken From: app_rpt-users-bounces(a)ohnosec.org [mailto:app_rpt-users-bounces@ohnosec.org] On Behalf Of Alan Adamson Sent: Tuesday, June 26, 2012 7:55 AM To: app_rpt-users(a)ohnosec.org Subject: [App_rpt-users] RTCM Question Anyone else noticed this? Maybe my user radio is just a fast switcher. I use and RTCM in conventional mode at my repeater to be the interface for it and Allstar. If I don't remember to keep the PTT held for about a second after I finish talking, as soon as I release the PTT, I get about 1 second (it's actually a smidge shorter than that) of the last few things I said on the air. I'm pretty sure this *repeat* is because of the approx 1 second delay that the RTCM instills in the system. It's about 400ms of startup delay, and about the same of system delays, so the live release of the PTT sees approx 800ms of delayed audio. If I remember to keep the PTT pressed for approx 1 second when I finish talking I don't see this. I'm using a commercial radio, so maybe I'll need to see about a programmed setting to do that for me automagically. Just wondering if others are seeing it as well or if I just have a fast switcher. Alan

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USB Keying
by Ryan Collier 26 Jun '12

26 Jun '12

Greetings, I have usb keying on one usb and rx audio and tx audio on another usb. Is there a way I can get the software to do this? I am having a hard time getting the fobs to work. So I have this working in my windows machine and wanted to see If I can transfer the setup to asterisk. Ryan

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Distorted audio - no explanation
by (KP4TR)Ramon Gonzalez 25 Jun '12

25 Jun '12

Hello, I have an interesting situation. I have been using Allstar for over a month discovering its capabilities. Tonite a friend on node 28679 tells me my audio is distorted. Im puzzled because I haven't changed any settings on my node in a few days, although I did setup a new server that will go next week to Puerto Rico. So I turn on a remoterig radio in Chicago (independent of Allstar) to listen to myself, and indeed my audio sounds awful. So I check for all possible causes, but nothing changes. I switched radios, power supplies, moved cables around, but problem continues. I even reset my network and changed switch. Yet I decided to use echolink to connect to same exact hardware/URI FOB etc, and audio sounds perfect. I thought it was my friends radio, but I connected his radio to the WINS network during a net, and audio was fine. So my question is this. My 2 nodes on same server can connect to each other (28532 -> 28689), audio sounds great. I connect to node in Chicago, and audio is awful. I use echolink with same exact hardware on both ends, audio is perfect. So it is not a hardware or audio setting problem. It seems to be network and/or related to VOIP audio. I have yet to test with someone else other then KP4EOP. I even tested with the new server I am setting up, and it also has same issues. Even changed ports. Anyone seen this happen?

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Compromise?
by Alan Adamson 24 Jun '12

24 Jun '12

Anyone know if there is a way to compromise on the telemetry options. I want the node to act like a simplex link with no CT or hang time, when it's keyed local, but when it's keyed remote (from another node), I want HT and CT or at least the option to one tweak them. Duplex=0 kills both but I need the flexibility to set either *or* both. I'm running on a digital repeater, that uses it's built in controller (no this isn't D-Star), I've figure out how to pick of the transcoded audio such that I can send it to allstar, but what I don't want is the HT and CT coming back to the repeater with the delay from allstar on local traffic through the repeater only. But, when a signal comes from allstar, from another link or node, I *do* want at lease the CT. I'll have to see if the HT on the local controller is still in effect. I've just about got my NXDN repeater functional as an NXDN to Allstar bridge (in fact that works today), without double transcoding the digital audio at the repeater (first test were to pick off the transcoded audio from the repeater RX, and let allstar duplex it back. well as you might imagine, lossy transcoded audio, then sent back in to be re-transcoded to a lossy process, to then be decoded by the user radio, didn't sounds so good :). ). Anyway, I'm all most there, just need this last piece of the puzzle. Thanks again, Alan

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Re: [App_rpt-users] Distorted audio - no explanation
by Alan Adamson 24 Jun '12

24 Jun '12

Ok, I *want* video!!! Doesn't allstar support that already? :).. Alan From: app_rpt-users-bounces(a)ohnosec.org [mailto:app_rpt-users-bounces@ohnosec.org] On Behalf Of Jim Duuuude Sent: Saturday, June 23, 2012 11:58 AM To: kp4tr.ramon(a)gmail.com Cc: app_rpt mailing list Subject: Re: [App_rpt-users] Distorted audio - no explanation Well, maybe we should get together, bring up a connection, and swap some tone. Please (off list, of course), send me your phone number and login info to your system (so that I can see it from both ends while we are testing). Time-wise today things are not looking good. It's definitely a majorly serious skater day. My friend and I are going to Westchester and also check out the old (rather bowly) park at Venice. So if I get back soon enough (keeping in mind the time difference there) I will call. However, realistically, we went to bed at about 0200 our time last night (er, this morning I guess). Jim _____ Date: Sat, 23 Jun 2012 11:06:24 -0400 From: kp4tr.ramon(a)gmail.com To: telesistant(a)hotmail.com CC: app_rpt-users(a)ohnosec.org Subject: Re: [App_rpt-users] Distorted audio - no explanation Yes, audio levels are set perfectly. Believe me, I am totally perplexed as to what is causing this as it just happened out of nowhere. I can connect from one node to another and audio sound great both ways. It's only when I connect outside my network using Allstar. If I use echolink from the one node that has it configured, to the node in Chicago I'm testing with, audio sounds great. Same hardware. It is not an audio level setting problem. It is not a usbradio option setting, as I use 2 entirely different radios and both have same situation. It is not a server setting, as I have tested on 2 separate servers, and same problem. I rebooted both servers several times. No change. The only common thing is my network and ISP. Tested with my IRLP node, all is ok. Tested bandwidth, everything looks good (average 40MBPS up/ 5MBPS down). On 6/23/2012 4:51 AM, Jim Duuuude wrote: have you set your audio levels to the standard app_rpt/Allstar TLP? Jim WB6NIL _____ Date: Sat, 23 Jun 2012 03:04:36 -0400 From: kp4tr.ramon(a)gmail.com To: app_rpt-users(a)ohnosec.org Subject: [App_rpt-users] Distorted audio - no explanation Hello, I have an interesting situation. I have been using Allstar for over a month discovering its capabilities. Tonite a friend on node 28679 tells me my audio is distorted. Im puzzled because I haven't changed any settings on my node in a few days, although I did setup a new server that will go next week to Puerto Rico. So I turn on a remoterig radio in Chicago (independent of Allstar) to listen to myself, and indeed my audio sounds awful. So I check for all possible causes, but nothing changes. I switched radios, power supplies, moved cables around, but problem continues. I even reset my network and changed switch. Yet I decided to use echolink to connect to same exact hardware/URI FOB etc, and audio sounds perfect. I thought it was my friends radio, but I connected his radio to the WINS network during a net, and audio was fine. So my question is this. My 2 nodes on same server can connect to each other (28532 -> 28689), audio sounds great. I connect to node in Chicago, and audio is awful. I use echolink with same exact hardware on both ends, audio is perfect. So it is not a hardware or audio setting problem. It seems to be network and/or related to VOIP audio. I have yet to test with someone else other then KP4EOP. I even tested with the new server I am setting up, and it also has same issues. Even changed ports. Anyone seen this happen? _______________________________________________ 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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DMK URI Help.
by Bradley Haney 24 Jun '12

24 Jun '12

Hello all .. still working on getting a cable correct connecting to a VERTEX 7000 repeater. i got some good info but needing a little more help.. ON Pin 8 of the DMK URI it says CTCSS_DET... I know what it says but i guess i am not quite sure on what it should be hooked to. Should it be hooked up to an OUTPUT of the repeater, input? Also got some good info from jim on setting the rpt.conf to link to link =yes and then duplex =0 since the repeater will be doing the repeating and ID, but what setting should i mess with so the auto patch uses full duplex.. IE it keys the repeater and keeps it key'd until the phone conversation is over.. Bradley

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