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

[A J Stiles]

On Wednesday 17 October 2012, bilal ghayyad wrote:
> Dears;
> 
> I am facing the following problem:
> 
> Already we requested from the service provider to enable the auto jumping
> service for our analoge telephone lines, so because we have 4 telephone
> lines from the service provider, then if you called line # 1 and it was
> busy, then the call will be sent to any available line #2 or #3 or #4, and
> if you call line # 3 and it was busy then the call will be sent via any
> available line of these four lines.
> 
> This feature is causing a problem at the Asterisk PBX, so some calls are
> not handled properly (it is ringing and we do not hear the welcome
> message), also the outgoing calls are facing a problem because it seems
> that there is a confusing happening in dahdi to determine the available
> line.
> 
> I do not know really how the automatic jumping feature is working at the
> service provider and what is the effecting on the DAHDI and Asterisk that
> is causing to not responding for the DAHDI channels properly.
> 
> For more details to be sure that I described the behaviour of the auto
> jumping feature that I took it from the service provider, let us assume my
> number is 22446789, when I call this number and I look for asterisk CLI, I
> can see that the call came via DAHDI/3-1 and then I do another call to
> this line, I can see it via DAHDI/4-1 and I do another call to this line
> and I will see it via DAHDI/2-1.
> 
> Also, not all my calls are failed ... but some are succeed and some are
> fails, so the responding is not perfect. I am sure because of the auto
> jumping feature from the service provider.

If you have multiple lines, and they are all paid for in the same name, then 
your telco really should have set it up so they are all accessible by dialling 
the same number.

Way back in the clicky-clicky days, having multiple lines connected to the 
same switchboard would have been done at the exchange by allocating sequential 
lines on the same selector, which was modified to step on until it found a non-
engaged line  (or go to engaged tone, if the last in the set were engaged).  
For instance, Radio Derby's main switchboard number was 361111; but 361112, 
361113, 361114, 361115 or 361116 might also reach the switchboard  (depending 
whether or not that line was already in use).

Digital exchanges don't have such requirements, of course; and since we went 
over to System X, which does not impose a 1:1 mapping between  (logical)  
numbers and  (physical)  lines, 361112  (at least)  has been allocated to 
another subscriber.  And there are many lines numbered 361111.

If you have several lines and they are properly grouped by the telco, you may 
get a call coming in via a differently-numbered line than what the other 
subscriber actually dialled.  

The way top deal with this in Asterisk is as follows:  Have one context that 
handles incoming calls from the PSTN  (usually  [from-pstn]  but you may have 
changed this).  In this context, you just need to handle calls for any 
extension the same.  (Or make sure, by using a catch-all such as the "s" 
extension or "_X.")

For calling out, make sure all your DAHDI channels are in the same group in 
chan_dahdi.conf, and use something in your Dial() command like 
Dial(DAHDI/g1/${EXTEN}) or Dial(DAHDI/r1/${EXTEN}) .  The "g" form will try 
always to use the lowest-numbered available channel; the "r" form will keep a 
track of which channel was used last and try to cycle through channels in turn 
from lowest to highest.  (Capital G1 and R1 will try always to use the highest 
number, and cycle through from high to low respectively).


-- 
AJS

Answers come *after* questions.