I would like to use the Asterisk Playback() command in extensions.conf before rpt() is called, or instead of calling rpt(). I'm finding that ASL3 disconnects after about 5 seconds, before the Playback() audio completes. I've looked for a setting that affects this time, but have not found one. Use of the Answer() command seems to make this playback connect time even shorter.
Perhaps related, I've found that the DVSwitch Mobile client allows the entire Playback() audio file(s) to be received from my ASL3 node, before or without rpt() being executed.
Is there an ASL3 setting that I can use to avoid disconnecting the node before Asterisk Playback completes?
The only settable timers that I know of near that
;lnkactenable=0
;lnkacttime=1800
Probably do not apply here because it has yet to connect. Timer has yet to start.
However, there is a connection attempt timer from the origination station, which I think runs till the connect message, but not a settable item I don't believe. I could be wrong. But you can't generally change others settings anyway.
Basically, you seem to be timing-out the originating connection attempt.
Ya know, you can say 3 times as much in a recorded sound compared to Allison's speech.
Or do it in 1/3 the time.
Perhaps show the stanza you are working with and I or someone here may figure out a alternative way of doing whatever it is that you are trying to accomplish.
If you are using "autopatch", and duplex is <3, there is a automatic "drop" time to allow for the operator to "control" the phone user. It's possible this is cutting your audio out.
If this is the case, it is controlled by:
[YourNode]
voxtimeout = 10000 ; VOX timeout time in ms
voxrecover = 2000 ; VOX recover time in ms
```How are you calling the dial plan?
There seems to be a hard coded timeout for rpt to be started on the remote node, after a node connects. This timeout appears to be about five seconds, or less, depending if the remote node uses asterisk answer() or not.
I want to use asterisk playback command(s) to send audio messages to connecting nodes lasting longer than this timeout when a node connects, but before rpt is started, on the answering node.
For example or demonstration, I have my 516227 node configured in extensions.conf to playback numbers from 1 through 10. If I enable asterisk to answer(), it gets to about 2. If asterisk does not answer, it gets to about 6 or 7.
If this timeout cannot be disabled or set today, could there be an enhancement added to optionally change or disable this rpt connect timeout?
(...)
same => n,Playback(digits/1&digits/2&digits/3&digits/4&digits/5&digits/6&digits/7&digits/8&digits/9&digits/10,noanswer)
(...)
or
(...)
same => n,Playback(digits/1,noanswer)
same => n,Playback(digits/2,noanswer)
same => n,Playback(digits/3,noanswer)
(...)
You can't control other nodes timers. It is not your timer but the connectee's timer.
They are timing out waiting for a connect message.
No setting for this.
There simply should be little delay after the inbound connection has passed past authorization to the actual connection. And that is what you are inserting, a delay.
You need to work within the limits you have.
I had done this with ACID &1.01 versions, never tried with v3 but I can tell you once again allson is slow in v3 and you are better off recording your own message for playback that is within the limits of the time available and you can convey more info in that short time frame. Which is likely something less than 7 seconds remaining depending on how long it took to get past authorization.. Which may vary based on the speed of your system and the congestion at that moment.
If you would like to share the whole stanza, I 'might' be able to help in some way.
If not, perhaps someone else can see in the dark better for help.
I don't know if this is possible. Even if it is, you'd have to change it on any remote node connecting to yours. This would also make things bad for a legitimate connection issue, I.E. congestion, wherein a node simply isn't responding.
My advice would be to make a very short recording (under 4.5 seconds long) stating whatever you want to be stated. Don't use Allison.
I did this on one of my hubs temporarily when it was having flaky performance, and I couldn't do anything about it for a couple of days. It was just a short file instructing people to connect to another node on my system instead. I made sure it was only about 3 seconds long, and nothing weird happened.
The nodes are connected, as far as Asterisk is concerned. It is Allstar RPT seems to insert an arbitrary timeout constraint that I'd like to avoid, or manage, before starting RPT on the remote end. I guess I'd like RPT to act more like an extension in Asterisk, insofar as timeouts.
An alternate enhancement suggestion might be to add a playback command to rpt that plays audio only to a specific connected node. At the moment, I"m limited to playback locally or to all RPT connections. It seemed like the better choice would be to just use the included Asterisk playback command before RPT is started. But that's not presently possible.
I have been playing the longer audio bulletin to all connected users using rpt playback, which is not what is desired.
Asterisk may be telling 'YOU' that it is connected to xxxxx because the process has been authorized, but xxxxx does not know it is connected to you until it hits this line in the stanza
same => n(connect),rpt(${EXTEN})
and the clock is tic'ing for them till it gets there.
Sorry you are not happy with timing-out a inbound connection.
That is on your code insertion causing a delay at a critical time..
Playback works just fine otherwise.
So, you want 'everyone' including those not connecting to you to have to wait more than a minute to find out if the connection will be successful so you can possibly play a message to inbounds before actual connection.
If you think that sounds interesting to other users ?
You can always post your request to the feature request page.
So, you want 'everyone' including those not connecting to you to have to wait more than a minute to find out if the connection will be successful so you can possibly play a message to inbounds before actual connection.
Nope, please re-read the topic subject. The keyword is/was configurable.
Asterisk provides the TIMEOUT(timeouttype) function, and still manages to time out or fall through call connections gracefully, under known rules. It seemed like there should be something like this for RPT connections, or use the Asterisk functions/features. I was wrong. mea culpa.
David, I am sorry once again you do not understand this, but you were answered 3 times that you can do this within limits.
The timer you want to adjust is on everyone else's system.
If you were to change your timer, it would only effect you when you were connecting 'outbound'.. The answer was no repeatedly.
If you had a timer adjustable on yours, it would 'not' change the outbound timer on connect of everyone else.
There is no timer on your system 'timing them out'. It is your hesitation/delay of sending them off to the connect code. If this were a pure asterisk thing you could do it without issue. But it is radio/app_rpt riding on asterisk and its rules apply when using it.
Everyone has a timer on the allowed time to try to connect. And until it sees the connect message, it continues to run. You could 'not' re-write your own software to prevent this.
It is not under your software's control.
If you still do not get this, I'm sorry. But Someone else will have to explain it.
All I tried to do is to keep you from wasting time trying to do something that you can do within limits and explained them. But I'm done. No thanks required.
Guess I am not a good explainer.