[asterisk-users] SIP Extension Config Issue

[Joseph L. Casale]

>exten => _1xxxxxxxxxx,1,Dial(SIP/200&SIP/201&SIP/202&SIP/203,30,tr)
>exten => _1xxxxxxxxxx,n,Voicemail(mailbox at vmcontext)
>
>Use whatever voice mailbox and voicemail context you want.

Well, its not advancing when *no* phones are online, just ringing busy.
It does however step through just fine when they *are* online.
I assumed that since it advances through correctly when they are online
there is something else that happens to asterisk when no peers inside are
registered.

Thanks for the help :)

jlc