[Asterisk-Users] Realtime problem

[Benjamin Stocker]

2006/6/22, Michiel van Baak <michiel at vanbaak.info>:
>
> On 16:57, Thu 22 Jun 06, Benjamin Stocker wrote:
> > Hi
> >
> > This works fine in extensions.conf:
> >
> > exten => _0X./100,1,Dial(SIP/${EXTEN}@sipout-a)
> > exten => _0X./200,1,Dial(SIP/${EXTEN}@sipout-a)
> >
> > This will just use different SIP channels for different Caller ID's.
> > If I write the same to a realtime table, Asterisk always uses sipout-a,
> no
> > matter what Caller ID is used.
>
> That will be the case with static configs too, because the
> argument to Dial is the same in both cases
>
>
That was a typo. Sorry, the second line reads:

exten => _0X./200,1,Dial(SIP/${EXTEN}@sipout-b)
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