[asterisk-users] SIP 484 (Early Dial) and International Dialing

[Olle E Johansson]

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 - oej at edvina.net
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