[asterisk-users] Solved !! Siemens Hicom --> Asterisk-Server <-- Telekom

Stefan Becker becker at yukonho.de
Fri May 27 15:05:17 CDT 2016

Original Subject: Questions... connecting Asterisk to the World

In short, adding the line 
 > overlapdial=yes
in chan_dahdi.conf changed everything!!

I wrote:
> WITH my Asterisks server in the middle of the exchange...
> A phone connected to the switch requests an "Outgoing" line
> by dialing "0". -->  Asterisks receives incoming call on "s".
> The dialed digits are collected. The dial plan is
> executed accordingly but the "caller" receives NO DIAL TONE
> and NO more information about the dialed number. The number is
> not placed in the "dialed" numbers simple functions like
> "redial" do not work anymore.

overlapdial=no (default behavior) Incoming extensions are being dropped.
Speed dialed numbers are incomplete.

;Span 1
;overlapdial=no   ;;or left out, default is 'no'
channel => 1-2

exten = _[0-9]XX.,1,Dial(SIP/${EXTEN}@dxt)

CLI >     -- Accepting call from '1713953' to '1361' on channel 0/1,
CLI >     -- span 1

last 2 digits missing - as if asterisk didn't wait long enough for them.
Extension '1361' is incomplete, the party actually dialed 136112 !!

CLI >     -- Accepting call from '136221' to '142629542' on channel 0/1,
CLI >     -- span 1

Party '142629542' is incomplete, the number actually dialed was
'9142629542' !!  The first digit is missing.

AND the dial tone is missing on the outgoing line.  There are alot of
"Work-a-Rounds" to simulate a missing Dial tone. Still - its missing !

First digit is missing as if asterisk wasn't fast enough to pick up the
dialed number. This always happens when using redial or speed dial -
when the number cames in fast.

The problem is NOT with the dialplan.  It's the signaling of the ISDN
and I had no Idea how to describe this adequately.

With all this already said and no solution from the list - it took me
a bit longer. I thought it had something to do with the ISDN signaling -
and it does.
I've read about others having a similar problem most often
people just miss the dial tone on the line. Now I can be happy to submit a
working solution.

A description can be found: 
difference between "ISDN call in En bloc or Overlap modes"

>"When configured for En bloc, the setup message should contain all
>necessary addressing information to route the call. In Overlap, the setup
>message does not contain the complete address. Additional information
>messages are required from the calling side to complete the called

Adding "overlapdial=yes" to the channel definition solved everything !!

;Span 1
channel => 1-2

NOW THE incoming calls complete correctly.....

CLI > -- Accepting overlap call from '16171223' to '1361' on channel 0/1,
CLI > -- span 1
CLI > -- Starting simple switch on 'DAHDI/i1/16171223-1'
CLI > -- Executing [136121 at NTBA:1] Dial("DAHDI/i1/16171223-1",
CLI > -- "DAHDI/g1/121") in new stack

With this little change "overlapdial=yes" the incomplete call is
accepted and with starting the "simple switch on.." is magically and
correctly collected and processed, then executing in the Dialplan.

I hope this might help someone else.  It took me a good part or 6 months
to come across this little and very effective configuration switch.

NOTE: I don't have anything to do with a Cisco routers/switch.
For me, the conflict I describe occurs between a:
Siemens Hicom (outgoing ISDN trunk) to a Asterisk server with a Digium BRI
And likewise:
from the Deutsche Telekom (incoming NTBA Anlagen Anschluss) to a Asterisk
server with a Digium BRI card.

my best regards,

Stefan Becker

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