I have been using app_rpt to initiate a call from my handheld radio to my SIP users.
Usually I do this in two ways, either I set an autopatch on PTT button press to directly call a pre-set extension ( by fixing a value in ivoxexten = 1001 and callonvox = 1, in rpt.conf);
Second method, I set a DTMF based dialing by setting *61{1001} command in rpt.conf, as shown below.
; Autopatch Commands
61 = autopatchup,noct = 1,farenddisconnect = 1,dialtime = 20000,context=Radio_1,quiet = 1 ; Autopatch up
62 = autopatchdn ; Autopatch down
And both of these have been working fine.
But, Now, I want to initiate a call ONLY when sound above a certain level is detected on Radio’s carrier detection. Is there any way to make my app_rpt DETECT audio being received on my RADIO via mic.
Here is my extensions.conf:
[Radio_1]
[Radio_1]
exten = _.,1,Set(CALLERID(num)=610)
exten = _.,n,SIPAddHeader(Call-Info: answer-after=1)
exten = _.,n,Goto(DLPN_Dialplan,${EXTEN},1)
You would need a input trigger point. And a way to read that trigger converted to execute.
The only thing I can think of is to use a external vox and tripping a input pin on the URI where you can pick that up for execution with script and dialplan action.
Perhaps using ON EVENT programming (wiki) or purely bash script.
The skinny is that there is no way (that I know of) to read the vox value in software.
But there is always more than one way to do things, but this is the first to come to mind… Others may get more complex.
Someone may have a better answer for you.
the thing is, i have a URI (usb to radio interface from DMK , link: https://dmkeng.com/ ) connected to my USB port, and in app_rpt.c i can see a “dovox” function which takes “sound_energy” & “noise_energy” as params for enabling vox.
now, i am not sure why this is not working. the function has been called twice inside rpt call function.
all i had to do was call “dovox” ensure speech_energy is above a threshold and then use it to initiate vox call by building psuedo channels.
All of that sagnoma software is very different.
I can’t begin to advise you on it other than to say it’s not a supported standard under app_rpt/ASL.
But the software maker has it’s own support forum. You should try that.
But if you want specific help with app_rpt/ASL,
tell us what version of the app_rpt/ASL software you are using.
[quote=“Mike, post:4, topic:22950”]
But the software maker has it’s own support forum. You should try that.
[/quote] can you provide me the link? all i have got are asterisk and asl support forums.
now, coming back to the version of software i am using is Asterisk 16, with app_rpt selected during “make menuselect” option.
i have to use this version bcoz i want to continue supporting my SIP users which were later deprecated.
As I said to you in the other thread, Asterisk 16 is NOT supported.
Neither of those are supported with with app_rpt. Current app_rpt requires Asterisk 20+ and DAHDI 3.2.0 for proper function. It is strongly suggested to use the Debian packages. You can also install from source if desired:
yes, i remember reading your response in other thread:
but i don’t get it, since, i am able to use exactly these versions of softwares and perfectly make radio calls via my DTMF-based call initiation and COR-based call initiation.
what do you mean when you say: [quote=“N8EI, post:6, topic:22950”]
Asterisk 16 is NOT supported
[/quote]
i am not using ASL, i am using asterisk and app_rpt that comes bundled by default.
my only issue, which i have not been able to solve yet, is to initiate a call on vox-enabling.
Where are you getting a source combination of Asterisk 16 with app_rpt? The app_rpt module has not been compatible with Asterisk since 1.4 until we released “ASL3” compatible with Asterisk 20. I fear you’re using some hacked-together source combo.