[Asterisk-Users] Realtime problem

[Benjamin Stocker]

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.
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