becker at yukonho.de
Guest
|
 Posted: Fri May 27, 2016 3:05 pm Post subject: [asterisk-users] Solved !! Siemens Hicom --> Asterisk-Ser |
 |
|
Original Subject: Questions... connecting Asterisk to the World
In short, adding the line
in chan_dahdi.conf changed everything!!
I wrote:
| Quote: | 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.
|
Without:
-----------
overlapdial=no (default behavior) Incoming extensions are being dropped.
Speed dialed numbers are incomplete.
....
;Span 1
context=NTBA
;overlapdial=no ;;or left out, default is 'no'
signalling=bri_cpe
channel => 1-2
....
....
[NTBA]
exten = _[0-9]XX.,1,Dial(SIP/${EXTEN}@dxt)
....
incoming:
-------------
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 !!
outgoing:
------------
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
first
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"
from:
http://www.cisco.com/c/en/us/support/docs/dial-access/
dial-on-demand-routing-ddr/23382-isdn-overlap-prob.html
quote:
| Quote: | "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
address."
|
Adding "overlapdial=yes" to the channel definition solved everything !!
....
;Span 1
context=NTBA
overlapdial=yes
signalling=bri_cpe
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@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
card.
And likewise:
from the Deutsche Telekom (incoming NTBA Anlagen Anschluss) to a Asterisk
server with a Digium BRI card.
my best regards,
Stefan Becker
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
Search
