David fire <[EMAIL PROTECTED]> > hi > in sip.conf there is a parameter calllimit or something like that use it...
I believe the SIP call-limit parameter drops the call if the call limit is exceeded and does not respond as if the phone were busy. Also, since I have different models of phones with different numbers of lines, I really do not want to do this manually for every extension I have. -- James > 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. _______________________________________________ -- 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
