[arm-allstar] Question 3

[Jeff Olinger]

Doug
The fix you gave me worked perfectly running the node-config.sh. i did 
not have to reboot asterisk to make it take affect. Hope a fix is able 
to be found shortly.
Jeff
W6jso

On 8/28/2015 5:49 AM, arm-allstar-request at hamvoip.org wrote:
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> Today's Topics:
>
>     1. Re: software update status (LaRoy McCann)
>     2. the variable audio delay (Ken)
>     3. Question (Jeff Olinger)
>     4. Re: Question (Doug Crompton)
>     5. Verified international source of 'hackable' sound	fob's
>        (Igor Videc)
>     6. Re: Question (Neil Sablatzky)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Thu, 27 Aug 2015 13:22:08 -0500
> From: LaRoy McCann <lmccann at dtisp.com>
> To: ARM Allstar <arm-allstar at hamvoip.org>
> Subject: Re: [arm-allstar] software update status
> Message-ID: <55DF5550.5030803 at dtisp.com>
> Content-Type: text/plain; charset=windows-1252; format=flowed
>
> David,
> Thanks for the update.
> Yeah, the audio delay fix is what I'm looking forward to.
>
> I know all about the family stuff coming before, as my wife calls, it
> "playing radio".
>
> Thanks to all of you that are making this possible.
>
> 73's
>
> LaRoy McCann, K5TW
>
>
> On 8/27/2015 11:33 AM, David McGough wrote:
>> Hi LaRoy,
>>
>> Progress has been slow in recent weeks, due primarily to family
>> commitments that I've got. At this point, the update is waiting on me.
>>
>> I am hopeful to minimally have a bug-fix interim release soon. The big
>> item on the list being resolution for the variable audio delay.
>>
>> I hope to have more details soon.
>>
>>
>> 73, David KB4FXC
>>
>>
>>
>> On Thu, 27 Aug 2015, LaRoy McCann wrote:
>>
>>> Doug,
>>> Just curious on how things are progressing with a new software update
>>> for the PI2?
>>>
>>> Thanks,
>>>
>>> LaRoy McCann, K5TW
>>>
>>>
>>>
>>>
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>>>
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>>>
>> _______________________________________________
>>
>> arm-allstar mailing list
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>>
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>>
>
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>
> ------------------------------
>
> Message: 2
> Date: Thu, 27 Aug 2015 17:46:21 -0400
> From: "Ken" <ke2n at cs.com>
> To: <arm-allstar at hamvoip.org>
> Subject: [arm-allstar] the variable audio delay
> Message-ID: <06b901d0e111$d003d4b0$700b7e10$@cs.com>
> Content-Type: text/plain; charset="us-ascii"
>
> I have been seeing references to this topic for some time.
>
> I had an experience a while back with something similar.
>
>   
>
> As an experiment I had increased the time delays on the jitter buffer (in
> iax.conf).  I found that (whatever was going on in my network) when the
> repeater cluster was not in use, the jitter buffer would wind up to its
> maximum extent. So it would always start out with this delay. I had 3000 ms
> for one of the parameters and I could say a lot of things before it would
> start coming out the other side (!)  Of course, it would shrink back down to
> nothing once things got going.   I think it was the resync threshold that it
> was up against.  Of course, the buffer itself must be larger than the resync
> threshold.
>
>   
>
> Since then I have tended to keep the buffer/threshold on the small side .
>
>   
>
> Ken
>
> KE2N
>
>   
>
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> ------------------------------
>
> Message: 3
> Date: Thu, 27 Aug 2015 18:29:18 -0700
> From: Jeff Olinger <jeffoli at sbcglobal.net>
> To: arm-allstar at hamvoip.org
> Subject: [arm-allstar] Question
> Message-ID: <55DFB96E.50707 at sbcglobal.net>
> Content-Type: text/plain; charset=utf-8; format=flowed
>
> I set my node up and it is up and running just fine but it is sending
> the voice id of wa3xyz with pl 88.5. During the setup I told it to do
> text to voice and I put my call sign in. Where can I go to correct this?
>
> Jeff
> W6JSO
>
>
> ------------------------------
>
> Message: 4
> Date: Thu, 27 Aug 2015 23:26:35 -0400
> From: Doug Crompton <doug at crompton.com>
> To: ARM Allstar <arm-allstar at hamvoip.org>
> Subject: Re: [arm-allstar] Question
> Message-ID: <BLU171-W55BB3A11ED63566AFD6EC5BA6E0 at phx.gbl>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Jeff,
>
>   Good that you brought this up because this applies to anyone setting up a node from the image. The Google TTS has stopped working due to a change in Google's terms of use. We will no longer be able to use it on either the BBB or RPi2. This mostly effects the weather scripts which depended on it heavily but is also used to setup the voice ID if you select TTS voice at setup.
>
> The easy way to fix it is to run the node setup again ( node-config.sh ) and hit enter through the selections as it will remember your last selection. When you get to the ID either select CW or simple gsm voice ID, NOT tts. Continue through the script to the end. Your ID should now be what you changed it to. You may need to do an astres.sh to restart Asterisk but I don't think that is necessary.
>
> The manual method is not hard but you have to understand how to edit and save files etc. If that is beyond the scope of you ability just use the simple way above.
>
> To do it manually you would first edit the /etc/asterisk/local/combine.sh  script. Edit it an change the call and whatever you want to say to your text. Remember you are combining Allison voices together so the words or letters have to be in the Asterisk sounds directories. Then run the script to create a custom gsm voice file. You then call this file from rpt.conf
>
> edit the /etc/asterisk/rpt.conf file. Modify the following lines -
>
> idrecording=/etc/asterisk/local/node-id
> ;idrecording=|iDE WA3XYZ/L
>
> This shows the playing of the voice ID from a file called node-id in /etc/asterisk/local
>
> If you commented that line out (; before it) and uncommented the following line and replaced the call with your call it would do a CW ID.
>
> Also if you are using a voice ID you can set the idtalkover parameter to switch to CW when a station keys up over your ID
>
> idtalkover=|iDE WA3XYZ/L
>
> Obviously change to your call. The CW ID's will send any text you enter.
>
> We are working on a possible fix for the loss of Google TTS but there is no great free solution for TTS in Linux. The tts feature will be removed from the script or modified in the next update.
>
> 73 Doug
> WA3DSP
> http://www.crompton.com/hamradio
>
>
>> To: arm-allstar at hamvoip.org
>> From: jeffoli at sbcglobal.net
>> Date: Thu, 27 Aug 2015 18:29:18 -0700
>> Subject: [arm-allstar] Question
>>
>> I set my node up and it is up and running just fine but it is sending
>> the voice id of wa3xyz with pl 88.5. During the setup I told it to do
>> text to voice and I put my call sign in. Where can I go to correct this?
>>
>> Jeff
>> W6JSO
>> _______________________________________________
>>
>> arm-allstar mailing list
>> arm-allstar at hamvoip.org
>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
>>
>> Visit the BBB and RPi2 web page - http://hamvoip.org
>>
>   		 	   		
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> ------------------------------
>
> Message: 5
> Date: Fri, 28 Aug 2015 10:09:16 +0200
> From: Igor Videc <9a6nvi at hamradio.hr>
> To: arm-allstar at hamvoip.org
> Subject: [arm-allstar] Verified international source of 'hackable'
> 	sound	fob's
> Message-ID:
> 	<CAL2m5yvX3nvvMALdGG-6xFacdZqxuidRY=XbWqwEgRu_bCxWtA at mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Hi,
> I have been following this list for some time now and as I see all of the
> suggested links to buy a compatible sound fob do not ship internationally
> (in my case Croatia).
>
> In 2012 I bought two "Music Fairy's"
> http://www.buyincoins.com/item/1325.html which have the normal CM108 chip
> and two weeks ago ordered two more but this time it came with a blob :(
>
> So my question is, do you know of a verified source (please send the link)
> of cmedia sound fobs that can be shipped internationally ?
>
> Thanks
> Igor
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> ------------------------------
>
> Message: 6
> Date: Fri, 28 Aug 2015 08:49:45 -0400
> From: Neil Sablatzky <k8it at cac.net>
> To: ARM Allstar <arm-allstar at hamvoip.org>
> Subject: Re: [arm-allstar] Question
> Message-ID: <117FBC57-1E6E-42EE-99C2-A8A95E62BEAC at cac.net>
> Content-Type: text/plain; charset="us-ascii"
>
> Doug As you are aware some Amateurs have purchased voices to get a high quality voice audio. Being blind it's a must to buy voices and software  much like buying a new handheld.  Please do not remove the TTS support as the feature works if an. Amater wants to spend dollars.  Maybe just remove the Google TTS option from the auto configure scripts.
> I do have a question about what script produces the GSM audio files.  What are the differences between GSM and the Allison files we can hand edit and Cat together?  Are they the same type of file but with a clever script That parcels the call sign ? If so what script is it and can it be used for  other messages?
> 73 Neil K8IT
>
> Sent from my iPhone
>
>> On Aug 27, 2015, at 11:26 PM, Doug Crompton <doug at crompton.com> wrote:
>>
>> Jeff,
>>
>>   Good that you brought this up because this applies to anyone setting up a node from the image. The Google TTS has stopped working due to a change in Google's terms of use. We will no longer be able to use it on either the BBB or RPi2. This mostly effects the weather scripts which depended on it heavily but is also used to setup the voice ID if you select TTS voice at setup.
>>
>> The easy way to fix it is to run the node setup again ( node-config.sh ) and hit enter through the selections as it will remember your last selection. When you get to the ID either select CW or simple gsm voice ID, NOT tts. Continue through the script to the end. Your ID should now be what you changed it to. You may need to do an astres.sh to restart Asterisk but I don't think that is necessary.
>>
>> The manual method is not hard but you have to understand how to edit and save files etc. If that is beyond the scope of you ability just use the simple way above.
>>
>> To do it manually you would first edit the /etc/asterisk/local/combine.sh  script. Edit it an change the call and whatever you want to say to your text. Remember you are combining Allison voices together so the words or letters have to be in the Asterisk sounds directories. Then run the script to create a custom gsm voice file. You then call this file from rpt.conf
>>
>> edit the /etc/asterisk/rpt.conf file. Modify the following lines -
>>
>> idrecording=/etc/asterisk/local/node-id
>> ;idrecording=|iDE WA3XYZ/L
>>
>> This shows the playing of the voice ID from a file called node-id in /etc/asterisk/local
>>
>> If you commented that line out (; before it) and uncommented the following line and replaced the call with your call it would do a CW ID.
>>
>> Also if you are using a voice ID you can set the idtalkover parameter to switch to CW when a station keys up over your ID
>>
>> idtalkover=|iDE WA3XYZ/L
>>
>> Obviously change to your call. The CW ID's will send any text you enter.
>>
>> We are working on a possible fix for the loss of Google TTS but there is no great free solution for TTS in Linux. The tts feature will be removed from the script or modified in the next update.
>>
>> 73 Doug
>> WA3DSP
>> http://www.crompton.com/hamradio
>>
>>
>>> To: arm-allstar at hamvoip.org
>>> From: jeffoli at sbcglobal.net
>>> Date: Thu, 27 Aug 2015 18:29:18 -0700
>>> Subject: [arm-allstar] Question
>>>
>>> I set my node up and it is up and running just fine but it is sending
>>> the voice id of wa3xyz with pl 88.5. During the setup I told it to do
>>> text to voice and I put my call sign in. Where can I go to correct this?
>>>
>>> Jeff
>>> W6JSO
>>> _______________________________________________
>>>
>>> arm-allstar mailing list
>>> arm-allstar at hamvoip.org
>>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
>>>
>>> Visit the BBB and RPi2 web page - http://hamvoip.org
>>>
>> _______________________________________________
>>
>> arm-allstar mailing list
>> arm-allstar at hamvoip.org
>> http://lists.hamvoip.org/cgi-bin/mailman/listinfo/arm-allstar
>>
>> Visit the BBB and RPi2 web page - http://hamvoip.org
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