Discussion: Asterisk cmd Flash

In my case to get call waiting working properly I had to use 'callee' instead of 'caller' in features.conf. When I tried it with 'caller' it attempted to Flash() the SIP line instead of the ZAP line and failing that, dropped the first call.

Also, putting DYNAMIC_FEATURES=>zapflash into the globals section of extensions.conf means you will not need to separately Set() it for any extra contexts you have.

extensions.conf 
[globals] 
DYNAMIC_FEATURES=>zapflash 

features.conf 
[applicationmap] 
zapflash => *0,callee,flash,() 

Hi!! i'm new in this.. i try to do flash with my pstn line, for use call waiting option but i try this command with trixbox 2.0 and not working.. i have a rhino card 4fxo .. plz send me back how i can do..

thantx

Carlos ventura
Carlos@gvr.com.do

I've solved the problem changing the flash time in the zapata.conf file,
I've set:

flash = 200 (the defualt was 750 ms)

in the extensions.conf the code is for example:

exten => 42,1,Flash()
exten => 42,2,SendDTMF(42,250)
exten => 42,3,Hangup()

now the transfer with flash works correctly

I've tried to transfer a call using the Flash command, but with my configuration it doesn't work.
I have a traditional PBX connected with a zap channel to Asterisk that acts like an IVR:

TELCO line --> traditional PBX (FXS) --> (FXO) Asterisk

From the TELCO line I can make a call to the traditional PBX and reach Asterisk, the IVR system on Asterisk answers the call and I can dial an extension (for example 42 that is on the traditional PBX). In the asterisk dialplan I've set to transfer the call using Flash() like in this example:

exten => 42,1,Flash()
exten => 42,2,Background(silence/1) wait 1 second for the traditional PBX
exten => 42,3,SendDTMF(42,250)
exten => 42,4,Background(silence/1) wait 1 second for the traditional PBX
exten => 42,5,Hangup()

When I dial the extension 42, the phone 42 on the traditional PBX rings but when I answer there isn't communication with the call from the TELCO line and after a few seconds the line hangup.
Here you can see what happen in asterisk CLI console:

Executing Answer("Zap/4-1", "") in new stack
— Executing BackGround("Zap/4-1", "a_suoni_plink/menu_esterno2") in new stack
— Playing 'a_suoni_plink/menu_esterno2' (language 'it')
== CDR updated on Zap/4-1
— Executing Flash("Zap/4-1", "") in new stack
— Flashed channel Zap/4-1
— Executing BackGround("Zap/4-1", "silence/1") in new stack
— Playing 'silence/1' (language 'it')
— Executing SendDTMF("Zap/4-1", "42") in new stack
— Executing BackGround("Zap/4-1", "silence/1") in new stack
— Playing 'silence/1' (language 'it')
— Executing Hangup("Zap/4-1", "") in new stack
== Spawn extension (incoming, 42, 5) exited non-zero on 'Zap/4-1'
— Hungup 'Zap/4-1'

I've tried the following changes to the dialplan in my example but transfer still doesn't work:

- I've tried to use wait(1) instead of Background(silence/1)

- I've tried without Background(silence/1) or wait(1):

exten => 42,1,Flash()
exten => 42,2,SendDTMF(42,250)
exten => 42,3,Hangup()

- I've tried without the Hangup() instructions at the end

Has anyone the same problem like me and any suggestions?

Hi,

I have everythng working on my * all the features that I want exept one.

I have a SPA-1000 connected to my home phones and X100P for the POTS line. The SPA gives me call waiting indication(beep) but when I press flash I just get dial tone. is it SPA-1000 thats doing something wrong or * not flashing the card ? Please help me with this issue because with out call waiting I cant use my * at all. Thank you.

Is there a way to simulate the transfer feature on a SIP line so it still only requires the one line for transferring a call, or is this functionality (three-way calling with transfer) not available on SIP lines?

How can we set the flash timing, and what is the default flash timing (in miliseconds) for Asterisk?

Can someone advice me on this?

Here is an alternative method that I just found. This makes it nice and simple to 'flash' an incoming POTS line (ZAP channel) as opposed to the dialplan scripts that I have seen that require tranferring the call, hanging up, and waiting for a call back. That was too confusing for my wife (me too). Now all that has to be done is press *3 and it is done. No transfers. No hanging up. No dial back.

extensions.conf
[context]
exten => s,n,Set(DYNAMIC_FEATURES=zapflash)

exten => s,n,Dial(SIP/xxxx,15,tw)

features.conf
[applicationmap]
zapflash => *3,caller,flash,() <<<<<<<<<< needs a comma between flash and ()

yah.. something like have it determine what phone is using the zap channel, then when you "transfer" the call, have it ring back to the phone that was originally using the zap channel instead of (in my case) ringing multiple phones...

Is there any way to utilize a second line of Sipura SPA-841 for CallWaiting?
When I'm on the first line (ZAP) and somebody else is trying to call my PSTN number I would see it as a call coming to the second line, so I can switch between two lines similar to FLASH on a regular phone.
Any ideas?