[asterisk-users] Automatic jump from line to line for incoming calls and the problem in DAHDI

A J Stiles asterisk_list at earthshod.co.uk
Thu Oct 18 03:51:35 CDT 2012


On Wednesday 17 October 2012, bilal ghayyad wrote:
> Actually I am not talking on how to handle it in the extensions.conf
> because I am doing same as you wrote. But even so, I am facing a problem
> that some calls are captured and some calls are not captured.
> 
> Currently, I set the callwaiting=no in the chan_dahdi.conf, it seems it is
> working fine. But I am not sure if this is really the required
> configuration to fix it or there is something else.
> 
> Any advise.
 
We need to determine the exact circumstances under which Asterisk is missing 
incoming calls, which is going to require some low-level hacking.

Wire up two LEDs back to back  (so whichever way the current is flowing, one of 
them will always light up).

*----->|-----*
*-----|<-----*

Put in series with this pair a 470 nF capacitor and a 100kΩ resistor:

*-----| |-----/\/\/\/-----[LEDs]-----*

When this contraption is connected across an analogue phone line, the LEDs 
will light up.  (One lights on the crest of the waveform and one on the 
trough, but this should be happening too fast for the eye to see, and it will 
just look like both are lit.)  The capacitor blocks DC on the line, so the 
LEDs will be off unless AC is present.  Make up 4 of these devices, so you can 
monitor the ringing status of each of the analogue lines going into your 
Asterisk box.  

In your extensions.conf, make sure that you have NoOp(${EXTEN}) somewhere in 
the [from-pstn] context, so you can see what number the upstream exchange was 
sending.  Be sure to include something which will keep a caller on the line 
for awhile.  Connect up a laptop with an SSH client to the network, so you 
have an Asterisk console *and* can see the ringing status of all 4 lines.

Lastly, you will need 4 mobile phones; and possibly volunteers to operate 
them, while you watch the Asterisk console and the LEDs.


Now you can investigate properly what is happening when you dial your analogue 
lines:

* Call a line which is not busy, by its own number.  Does Asterisk respond to 
the call?
* Call a line which *is* busy, by its own number.  Does the call automatically 
appear on another line?  Does Asterisk respond to it?
* Call a non-busy line while another line is busy.  Does Asterisk respond to 
the call?


You should be able to work out eventually just what is causing Asterisk to 
miss calls.


-- 
AJS

Answers come *after* questions.



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