[Asterisk-Users] Realtime problem

[Michiel van Baak]

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


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Michiel van Baak
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