2 apr 2007 kl. 22.21 skrev James FitzGibbon:
I'm building a dialplan for use with a bunch of GXP2000 desk sets.
During testing, we had some user issues surrounding the lack of an
on-phone dialplan. Users would hit 9 and sit there waiting for a
redial tone, and the GXP would time out, sending just '9' to *,
which couldn't do much other than spit back a 404 or play pbx-invalid.
I turned on the "early dial" option on the GXP, which causes each
digit to be sent as it is pressed, and the user response was much
more favourable. Now I come to set up my international dialplans
and I'm running into a problem.
The textbook dial pattern for international calls:
_9011.
Isn't working because * matches the first digit after 011 and sends
an incomplete dialstring (dialing something like Zap/R1/0119 for
example).
I've tried using patterns with multiple . wildcards, and switching
from . to X, putting patterns like
_9011XXX
_9011XX
_9011X
In the hopes that * would see that "90119" could potentially match
a longer extension and not match immediately. No luck though -
dialing still starts immediately when one digit past 011 is received.
Any thoughts on how to get around this? Right now the best I have
(and that's not saying much) is to have something like:
[initialcontext]
exten => _9011,1,DISA(no-password|somecontext)
[somecontext]
exten => _X.,1,Dial(Zap/R1/011${EXTEN})
But that's ugly, not to mention confusing to the users because the
amplitude of the dialtone generated by the GXP is lower than the
dialtone generated by *, so they notice the bump when they've
dialed 9011.
When SIP sends an INVITE, it's a complete INVITE. The dialstring in
the invite is "done" and can't be added to, unless you have
enabled overlap dialling in SIP. When the phone sends a number, we
match and set up the call or fail.
Overlap dialling in SIP works by testing the dialstring. If it's not
an exact match, Asterisk will send a SIP response saying that
it needs more digits to determine the destination. In 1.4, this is
disabled by default and needs to be enabled.
You are assuming that SIP works like zaptel in the dialplan, but it
does not. You propably need to re-configure your phones.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP master class, Stockholm may 2007 - register now!
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