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becker at yukonho.de
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 Posted: Sat May 14, 2016 11:51 am Post subject: [asterisk-users] Questions... connecting Asterisk to the Wor |
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Greetings,
asterisk list and community,
I have a problem in how our telefon switch (Siemens HiCOM)
"talks" with my new configured Asterisk server (V.11.18.0)
without my Asterisks server in the middle....
<phone> <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom
A phone connected to the switch requests an "Outgoing" line
by dialing "0". The party is connected via ISDN to
the carrier (deutsche Telekom) where the party preceeds
to dial numbers... and the call is connected....
What I can see while I am dialing is that with every
digit I press it is being displayed on my phone.
Further more, these digits are being processed by the
carrier. The call goes through, rings, immediately on
completion on the number or is rejected if busy.
WITH my Asterisks server in the middle of the exchange...
A phone connected to the switch requests an "Outgoing" line
by dialing "0". --> Asterisks recieves incoming call on "s".
The dialed digits are collected. The dial plan is
executed accordingly but the "caller" recieves no
more information about the dialed number. The number is
not placed in the "dialed" numbers simple functions like
"redial" do not work anymore.
Does anybody know what I am doing wrong here. Is there a
way to teach asterisk to behave exactly as if it were the
PBX (deutsche Telekom).
So, as to say, act in a way that NO ONE will rightly know
the differance between having asterisk taking over the
function of the ISDN PBX.
What do I need? A better dial plan to somehow better simulate
the way the switch normaly behaves?
Is hardware the problem?
My ISDN card in the server is:
"QuadBRI ISDN Digium Wildcard b410P"
Most everything else functionly works. incoming and outgoing calls
from and to ISDN, VoIP and other equipment work fine.
Just that the phones and switch don't recieve the "collected"
number sequence the was dialed.
Any help or ideas anyone might have would be greatly appreciated.
thanks,
Stefan
--
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asterisk.org at sedwar...
Guest
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| Posted: Sat May 14, 2016 12:39 pm Post subject: [asterisk-users] Questions... connecting Asterisk to the Wor |
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On Sat, 14 May 2016, Stefan Becker wrote:
| Quote: | A phone connected to the switch requests an "Outgoing" line by dialing
"0". --> Asterisks recieves incoming call on "s". The dialed digits are
collected. The dial plan is executed accordingly but the "caller"
recieves no more information about the dialed number. The number is not
placed in the "dialed" numbers simple functions like "redial" do not
work anymore.
|
This is not my area of expertise, but I'll throw my $0.02 in...
When you 'request an outgoing line' by dialing 0, that call leg is
processed by Asterisk, thus, that is what the phone 'sees' as the dialed
number and that's what the phone will send when 'redial' is pressed.
I think you need to make the outbound dial a single 'transaction' either
by using an extension pattern that includes the 0 like '05555555555' to
dial 555-555-5555 or eliminate the 0 (and the idiom of 'requesting an
outgoing line') and detect an internal vs external call via extension
pattern matching.
Does your internal extension numbering plan conflict with external
national numbering plan?
Dialing a prefix digit seems so 1970s to me.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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becker at yukonho.de
Guest
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| Posted: Sat May 14, 2016 2:32 pm Post subject: [asterisk-users] Questions... connecting Asterisk to the Wor |
|
|
On Sat, 14 May 2016, Steve Edwards wrote:
| Quote: | I think you need to make the outbound dial a single 'transaction' either by
using an extension pattern that includes the 0 like '05555555555' to dial
555-555-5555 or eliminate the 0 (and the idiom of 'requesting an outgoing
line') and detect an internal vs external call via extension pattern matching.
|
this is the dialplan that I use:
[ReceiveCallOut]
exten = s,1,Read(LOKAL,,,,1,5)
same = n,Dial(SIP/${LOKAL}@tt)
same = n,Hangup()
When the user dials "0", the HiCOM ISDN switch immediately
goes "online" to the outgoing ISDN Copper Cable - connected
to ... A) .... B)
A) connected to the NTBA in the wall jack to the NTBA phone company...
the dialing preceeds to continue "offline" no dailtones are heard.
The call is completed and connects
B) connected to the Asterisk ISDN Card....
Asterisk server reacts by executing the above dial plan...
CLI > "answered call from "...." to "s"
The user has an open "answered" line and the dialing are collected by
listening to the DTMF tones. The generated dial tones can be heard
on the phone line.
Somehow the signaling on the line of the outgoing call is differant
when the cable is handeled by the PBX or by asterisk.
But why ?
Can't asterisk be configured to handle a call exactly as the
otherwise connected phone company's PBX would?
thanks for listening,
Stefan
--
_____________________________________________________________________
-- 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 |
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asterisk.org at sedwar...
Guest
|
| Posted: Sat May 14, 2016 9:42 pm Post subject: [asterisk-users] Questions... connecting Asterisk to the Wor |
|
|
On Sat, 14 May 2016, Stefan Becker wrote:
| Quote: | On Sat, 14 May 2016, Steve Edwards wrote:
| Quote: | I think you need to make the outbound dial a single 'transaction'
either by using an extension pattern that includes the 0 like
'05555555555' to dial 555-555-5555 or eliminate the 0 (and the idiom of
'requesting an outgoing line') and detect an internal vs external call
via extension pattern matching.
|
this is the dialplan that I use:
[ReceiveCallOut]
exten = s,1,Read(LOKAL,,,,1,5)
same = n,Dial(SIP/${LOKAL}@tt)
same = n,Hangup()
|
I would:
) Drop the 'dial 0' anachronism.
) Not use read().
) Use extension pattern matching.
For example, in the US, I would have something like (off the top of my head):
; external, local
exten = _nxxxxxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n, goto(dial-local,${EXTEN},1)
same = n, hangup()
; external, domestic
exten = _nxxnxxxxxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n, goto(${CONTEXT},1${EXTEN},1)
same = n, hangup()
; external, domestic
exten = _1nxxnxxxxxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n, goto(dial-domestic,${EXTEN},1)
same = n, hangup()
; international
exten = _011x.,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n, goto(dial-international,${EXTEN},1)
same = n, hangup()
; internal
exten = _[2-9]xxx,1, verbose(1,[${EXTEN}@${CONTEXT})
same = n, goto(dial-internal,${EXTEN},1)
same = n, hangup()
| Quote: | When the user dials "0", the HiCOM ISDN switch immediately
goes "online" to the outgoing ISDN Copper Cable - connected
to ... A) .... B)
A) connected to the NTBA in the wall jack to the NTBA phone company...
the dialing preceeds to continue "offline" no dailtones are heard.
The call is completed and connects
|
This sounds weird and very foreign (strange and unfamiliar, not as being a
characteristic of a different country) to me. So, as a caller, I would
hear the '0' DTMF but no other tones? No feedback as I press keys?
| Quote: | B) connected to the Asterisk ISDN Card....
Asterisk server reacts by executing the above dial plan...
|
The dialplan does not reflect your intentions.
| Quote: | CLI > "answered call from "...." to "s"
The user has an open "answered" line and the dialing are collected by
listening to the DTMF tones. The generated dial tones can be heard
on the phone line.
Somehow the signaling on the line of the outgoing call is differant
when the cable is handeled by the PBX or by asterisk.
But why ?
|
Dialplan and channel configuration.
| Quote: | Can't asterisk be configured to handle a call exactly as the otherwise
connected phone company's PBX would?
|
My guess is yes.
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
https://www.linkedin.com/in/steve-edwards-4244281
--
_____________________________________________________________________
-- 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 |
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asterisk_list at earth...
Guest
|
| Posted: Mon May 16, 2016 3:10 am Post subject: [asterisk-users] Questions... connecting Asterisk to the Wor |
|
|
On Saturday 14 May 2016, Stefan Becker wrote:
| Quote: | Greetings,
asterisk list and community,
I have a problem in how our telefon switch (Siemens HiCOM)
"talks" with my new configured Asterisk server (V.11.18.0)
without my Asterisks server in the middle....
<phone> <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom
A phone connected to the switch requests an "Outgoing" line
by dialing "0". The party is connected via ISDN to
the carrier (deutsche Telekom) where the party preceeds
to dial numbers... and the call is connected....
What I can see while I am dialing is that with every
digit I press it is being displayed on my phone.
Further more, these digits are being processed by the
carrier. The call goes through, rings, immediately on
completion on the number or is rejected if busy.
WITH my Asterisks server in the middle of the exchange...
A phone connected to the switch requests an "Outgoing" line
by dialing "0". --> Asterisks recieves incoming call on "s".
The dialed digits are collected. The dial plan is
executed accordingly but the "caller" recieves no
more information about the dialed number. The number is
not placed in the "dialed" numbers simple functions like
"redial" do not work anymore.
Does anybody know what I am doing wrong here. Is there a
way to teach asterisk to behave exactly as if it were the
PBX (deutsche Telekom).
So, as to say, act in a way that NO ONE will rightly know
the differance between having asterisk taking over the
function of the ISDN PBX.
What do I need? A better dial plan to somehow better simulate
the way the switch normaly behaves?
Is hardware the problem?
My ISDN card in the server is:
"QuadBRI ISDN Digium Wildcard b410P"
Most everything else functionly works. incoming and outgoing calls
from and to ISDN, VoIP and other equipment work fine.
Just that the phones and switch don't recieve the "collected"
number sequence the was dialed.
Any help or ideas anyone might have would be greatly appreciated.
|
Your problem is that you are still thinking in terms of old-fashioned, clicky-
clicky mechanical telephone exchanges. Instead of "dialling 0 to request an
outside line", you need to let Asterisk accept all the digits and then
determine for itself whether the call is going to be an inside or outside one.
- If the user dials 3 digits (or however long your internal numbers are),
treat it as an internal number.
- If the user dials 6 digits (or however long numbers are on your local
exchange), treat it as an external, local number.
- If the user dials 11 digits starting with 0 (or however long a number is in
your country, including the STD code), treat it as an external, STD number.
- If the number dials 9 or more digits starting with 00, treat it as an
external, IDD number.
It will make your dialplan a little more complicated; but if it is too simple,
you won't be taking full advantage of the power of Asterisk.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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 |
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