hi in sip.conf there is a parameter calllimit or something like that use it... David
2008/12/5 James Lamanna <[EMAIL PROTECTED]> > Hi, > > I've noticed that if I have a multi-line linksys (942 or 962) phone > with the same sip registration mapped to each line key, that if all > the lines are full the phone will accept another call. I would expect > the phone to respond with "busy" so the call would to directly to > voicemail. > > Has anyone else experienced this and know of a workaround? I know it > seems like an endpoint issue and not an asterisk one. > > Thanks. > > --James > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination.
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