March 2017 - App_rpt-users - lists.keekles.org

allstar on dial
by K7ILO 14 Mar '17

14 Mar '17

Hello gang. So I decided to switch over to dial from acid so that my system can stay being supported so I installed dial onto a PC based machine. All the articles I have read by doing a Google search says that allstar will start downloading and installing after a few seconds of dial installation. Well, mine hasn’t started in 2 days so is there a script somewhere that needs to be started? Im sure this question has popped up but I have forgotten or do not know how to retrieve messages from the archives. Could someone point me in the direction of getting allstar to download and install so that I can set my system back up. Thanks K7ILO Las Vegas, NV

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Simulcasting with RTCM/voter
by Hayden Honeywood 14 Mar '17

14 Mar '17

Hi All, Thanks for all on the list who have given me a hand with getting the RTCM/voter boards running. I currently have 1/3 of my system installed at the sites. It's as follows: A TX only site - running a VOTER board, latest "Chuck" firmware. 9.6MHz OCXO replacing the onboard crystal and a 10MHz OCXO to keep the transmitter stable all on 53MHz. On my bench at home I have an identical setup except with a receiver in addition. Both radios are the same, audio injected in the same spot. I've been testing various frequency offsets to avoid the cancelling of the signal due to the inherent propagation effects at 6 metres. I've been able to run my bench radio into a dummy load and place it so that another radio nearby connected to an external antenna is receiving them almost the same (i.e. overlap). Currently 1-2Hz sounds the best. I have been encoding tone, however there is a very audible hum, as if the CTCSS tone from each transmitter is not in phase. The tone is 91.5Hz. I've checked/readjusted deviation levels etc, also checked the 9.6MHz OCXO etc but I cannot seem to null it out. I've also tested the Simulcast Launch Delay.The main site is about 13 miles away, so I put a value of 330 in there to represent a delay of 66uS on the bench test radio. This makes a little bit of improvement, but not an overall huge improvement. Does anyone have an ideas that I've missed or more information on the usage of the Simulcast Launch delay? Hayden

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Simulcasting with RTCM/voter
by Hayden Honeywood 14 Mar '17

14 Mar '17

Thanks for the information Joe. The RTCM generates the CTCSS in the main audio path with an entry in voter.conf. There is no separate deviation adjustment on the board, you set a global level in that config file. OK on the delays between sites etc. I think I understand whats going on there. As I mentioned beforehand I think a lot of my issue is that the 2nd site radio is on the bench and the delay from the main site TX is causing distortion problems. I'm hoping to document a lot of the simulcast information on my blog in time. In answer to some of your points - 1) I've played with the offset mainly because at low band 50MHz due to the propagation effects, complete cancellations happen in overlap when both signals from the transmitters are approx the same. Whereas on UHF I believe this is not to much of an issue due to reflections, doppler etc. Only really an issue with fixed stations I believe. Offsetting by a few Hz I can hear the null move around, and cancellation is not so bad. 2) The exciters have been set the same - same model, same inputs etc. 3) Yes. The only differences may be is that the VCO of the units is modified to 50MHz from 70 by adding capacitance in various places. The alignment of both VCO's is the same however - set to 3V on a specific test point. 4) Yep all the same. I've used a CTCSS encoder input which is post pre-emphasis and limiting. Limiting is done by the RTCM. 5) As above - although I've noticed that the CTCSS's deviation on the bench test transmitter is about 50Hz low when aligning for a 1K = 3K deviation. I've tried turning up the TX level pot to equal it, but it still sounds the same. Hayden

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DTMF timing
by mca555＠mac.com 12 Mar '17

12 Mar '17

Hi all, I’m having an issue with certain portable radios where the DTMF tones seem to be ‘bouncing’, i.e. a single tone is decoded multiple times for one press. The simple answer is “Buy a better radio!”, but that’s way too easy. The question is: Is there a way to tell the DTMF decoder to require a digit to be held longer before decoding? Thanks! Mark

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Re: [App_rpt-users] RPi w/DIAL Repeater Stuck In Transmit Mode
by DuaneVT . 11 Mar '17

11 Mar '17

I just bought an RA-35 to replace my node's DIY fob. VERY NICE BUILD! I should have gone with the RA-35 when considering the futzing to wire the fob and several unuseable fobs with blob ICs. Duane KA1LM 42996

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Certificate for allstarlink.org
by Steve Zingman 10 Mar '17

10 Mar '17

Tim has updated the certificate for the ASL web site. All browsers should be happy again. "Happy browser, happy life?" Steve N4IRS -- "Real programmers us vi, that's why I use nano"

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Re: [App_rpt-users] Codecs
by Benjamin L. Naber 08 Mar '17

08 Mar '17

Interesting to learn some of the workings of iax.conf. I'm aware of the rest, however, I would encourage iax.conf is modified to have the following priorities for every connection stanza: ulaw g726aal2 gsm this is to ensure audio quality is better than IRLP and echo-stink. If you cannot tell the difference, listen to a conversation on a local repeater, and the same conversation with the convenient delay over IRLP or echo-stink. You may then remember in days-past of analog cellular listening to the person on the other end had more body and fidelity. Regarding bandwidth, be it a DSL or mobile connection, this should be a non-issue. Even 1G cellular data connection will support two full-duplex uLaw streams. For hub nodes, 30 ulaw streams is right around 2.25Mbits. If this is a concern, perhaps the owner of the hub should really consider putting the node elsewhere. That's like trying to put a Pinto engine in my E-350 and expecting it to work the same. Considering jitter and packet loss if it is thought a higher bandwidth codec will cause this, such is not the case. This is the cause of little or no implementation of Qos/ToS in the WAN facing side of a router. There are other external considerations, but those are outside the scope of this thread. While on that subject a brief description of an example is the node at my home. My connection, fortunately, has maintained +25Mbps down and 5Mbps up, I've have zero issues with jitter and packet loss with QoS implemented, even while the kids are watching netflix, skyping, and watching something on youtube.... all at the same time. On 03/06/2017 11:08 PM, David McGough wrote: > Hi Benjamin, > > If you want to only use the G.711 CODEC, only allow that one CODEC at your > end. But, be aware that may not be compatible with all hubs out there. > Also, be aware that if you do NOT run the same CODEC as that which is > preferred by a given hub, you'll likely end up with poorer audio due to > transcoding. > > If you only want G.711, set your config such as: > > [whatever_stanza] > disallow=all > allow=ulaw > > > ....On a side note, while I agree with you that the Asterisk GSM and speex > CODECs presents lower quality audio, with many others (such as G.726 and > iLBC), you would be hard pressed to tell the difference between them, with > repeater audio. If fact, I'm not sure _I_ could tell the difference in a > blind A/B test. > > Another factor to consider is the hub bandwidth used for EACH audio > stream. This is a BIG deal for hub out-bound bandwidth. Here are results > from actual measurements on my hub: > > One IAX2 half-duplex stream. All results in Kbytes-per-second: > > GSM 3.2 > iLBC 3.5 > G726 5.5 > ULAW 9.4 > > > These data rates will be in-bound or out-bound, depending of if your local > node is listening or talking to another node across the VoIP network. If > you have 30 nodes connected to a hub, the total bandwidth required is 31 > times the above numbers---30 out-bound listeners + 1 in-bound talker. > > 73, David KB4FXC > > > > On Mon, 6 Mar 2017, Benjamin L. Naber wrote: > >> While reading an email from another VoIP email reflector, I got to > thinking about some of the codecs we use. > > Perusing through the iax.conf through a few allstar network linking > distros, each seem to have their own preferred set of codecs. > > I'm rather disturbed that iLBC and speex remains in any of them, while > I'm sure for compatibility of older installations. However, low quality > bandwidth codes sounds no better than echo-stink. This is one of the > reason why we use Allstarlink, for better audio quality, is it not? > > As I connect to different allstar nodes throughout the world, more-so in > the US, some are using iLBC or gsm... and a majority are hub nodes. This > to me, is unacceptable, and appears the codec being used are whatever > the allstar box has been packaged with. > > So, given than I'm not a Steve Henke or Steve Zingman, but I want to > ensure that I'm doing my part to achieve high fidelity through g711 > codec and proper URI <-> radio hardware interfacing techniques. > > Which stanza in the iax.conf is for out-going connections? > > I was under the impression [radio-proxy] was, however, when a node with: > disallow=all > allow=ulaw > allow=g726aal2 > allow=gsm > > connects to my node which has the *same* key=value pairs under [radio] > stanza, I got the message something along the lines of: > > connected: IP > codes available: g726aal2/gsm > host codecs available: ulaw|g726aal2|gsm|ilbc > actual codec: g726aal > ... > > bost sides do not have ilbc *anywhere*. The codec has been > removed/deleted. Why does it still show up? > > What is the ultimate deciding factor of which codec is used if both side > have codecpriority=mine? > > ~Benjamin, KB9LFZ > > _______________________________________________ > App_rpt-users mailing list > App_rpt-users(a)lists.allstarlink.org > http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users > > To unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the "Unsubscribe or edit options button" > You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem. >

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WebTransceiver - just me?
by DuaneVT . 07 Mar '17

07 Mar '17

While EVERYONE says WT does not work, I have it working on several Win7 32 bit computers. Sometimes I don't get the Java block running, but for the most part, failures are the receive audio does not show on the Windows speaker mixer panel. Other times, the level bar bounces with audio but nothing is heard on the computer speakers. WHEN it works, it works nicely. 73, Duane KA1LM 42996

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Re: [App_rpt-users] App_rpt-users Digest, Vol 97, Issue 9
by no1pc＠yahoo.com 07 Mar '17

07 Mar '17

To be clear, I have no problem getting WebTransceiver setup and functional... ... what I experienced seemed to be a glitch or lack of connection from the host/server process to either of 3 nodes that had worked fine a couple of days prior. Could have been a 'Murphy' day. I haven't had time to re-check. 1. Re: WebTransceiver - just me? (k5mvz(a)verizon.net) ---------------------------------------------------------------------- Message: 1 Date: Tue, 7 Mar 2017 11:16:29 -0500 From: k5mvz(a)verizon.net To: app_rpt-users(a)lists.allstarlink.org Subject: Re: [App_rpt-users] WebTransceiver - just me? Message-ID: <15aa98ebf97-43df-1801f(a)webprd-m11.mail.aol.com> Content-Type: text/plain; charset="utf-8" Duane- I have been running Web Transceiver using Linux Mint/Mint xfce/Zorin/Ubuntu & Firefox Web Browser for 2-3 years & it seems to work ok. This is my primary & only access to Allstar. I also have Web Transceiver running under Win XP & Firefox with no known problem, but not a lot of runtime. I also have it on a laptop running Win 7 & Firefox & it seems to work ok, but not a lot of runtime as well. Had it on the Win 7 laptop for about a year. I primarily run Web Transceiver using Linux & Firefox at home & on a laptop when traveling. Since U are running Win 7, U might want to run Firefox or Chrome for a Web Browser. This may help the the audio problem. U did not say what web browser U were running with Win 7. This came up a year or so ago on the forum. It may have been U looking for an alternative for computer users in ur group/club. Hope this helps. de- Bob K5MVZ -----Original Message----- From: DuaneVT . <selkie2(a)comcast.net> To: app_rpt-users <app_rpt-users(a)lists.allstarlink.org> Sent: Mon, Mar 6, 2017 12:20 pm Subject: [App_rpt-users] WebTransceiver - just me? While EVERYONE says WT does not work, I have it working on several Win7 32 bit computers. Sometimes I don't get the Java block running, but for the most part, failures are the receive audio does not show on the Windows speaker mixer panel. Other times, the level bar bounces with audio but nothing is heard on the computer speakers. WHEN it works, it works nicely. 73, Duane KA1LM 42996 _______________________________________________App_rpt-users mailing listApp_rpt-users@lists.allstarlink.orghttp://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-usersTo unsubscribe from this list please visit http://lists.allstarlink.org/cgi-bin/mailman/listinfo/app_rpt-users and scroll down to the bottom of the page. Enter your email address and press the "Unsubscribe or edit options button"You do not need a password to unsubscribe, you can do it via email confirmation. If you have trouble unsubscribing, please send a message to the list detailing the problem.

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Re: [App_rpt-users] Better Telemetry!!!
by Skyler F 07 Mar '17

07 Mar '17

Didn't finish typing before I hit send. Once the editing is done, in the asterisk directory, type ./configure make make install If anybody has any comments on how to add it as an option in the rpt.conf file, let me know. The code is pretty dirty at this stage, and it is my first real modification to the source code. On Tue, Feb 28, 2017 at 11:30 PM, Skyler F <electricity440(a)gmail.com> wrote: > I was toying around in app_rpt.c, and I figured out a way to shorten the > telemetry down. > > NODE XXXXX Connected to node XXXXX > > Is now replaced with: > > NODE XXXXX Connected > > > First, go to the app_rpt.c file, located in > /usr/src/utils/astsrc/asterisk/apps > > OR > /usr/src/astsrc-1.4.23/asterisk/apps > > > View the file app_rpt.c and search for this code below: > > > if (n < 3) return; > if (wait_interval(myrpt, DLY_TELEM, mychannel) == -1) > return; > res = saynode(myrpt,mychannel,strs[2]); > if (!res) > res = ast_streamfile(mychannel, "rpt/connected", > mychannel->language); > if (!res) > res = ast_waitstream(mychannel, ""); > else > ast_log(LOG_WARNING, "ast_streamfile failed on > %s\n", mychannel->name); > ast_stopstream(mychannel); > res = ast_streamfile(mychannel, "digits/2", > mychannel->language); > if (!res) > res = ast_waitstream(mychannel, ""); > else > ast_log(LOG_WARNING, "ast_streamfile failed on > %s\n", mychannel->name); > ast_stopstream(mychannel); > saynode(myrpt,mychannel,strs[1]); > return; > > > > Replace That chunk of code with : > > if (n < 3) return; > if (wait_interval(myrpt, DLY_TELEM, mychannel) == -1) return; > if (strcmp(strs[1], myrpt->name) == 0) > { > ast_verbose("\nDialed OUT\n"); > res = saynode(myrpt,mychannel,strs[2]); > } > else{ > ast_verbose("\nDialed IN\n"); > res = saynode(myrpt,mychannel,strs[1]); > } > ast_verbose("Node Connection Test strs[1]=%s ,strs[2]=%s, > my_node=%s\n",strs[1],strs[2], myrpt->name); > if (!res) > res = ast_streamfile(mychannel, "rpt/connected", > mychannel->language); > if (!res) > res = ast_waitstream(mychannel, ""); > else > ast_log(LOG_WARNING, "ast_streamfile failed on > %s\n", mychannel->name); > ast_stopstream(mychannel); > /*res = ast_streamfile(mychannel, "digits/2", > mychannel->language);*/ > if (!res) > res = ast_waitstream(mychannel, ""); > else > ast_log(LOG_WARNING, "ast_streamfile failed on > %s\n", mychannel->name); > ast_stopstream(mychannel); > /*saynode(myrpt,mychannel,strs[1]);*/ > return; > -- Skyler Fennell amsatnet.info KDØWHB electricity440(a)gmail.com

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