On Sunday 22 July 2007 21:20, Nicholas Blasgen wrote: > I've been having some problems recently with Asterisk thinking a SIP phone > is still connected. These are GrandStream Budge Tone 102's and even after > someone hangs up the AGI script I have running for them is still looping.
FWIW, we set up bt102's as not needing registration i.e. we set host=<actual ip> instead of dynamic and configured the phone to allow calls without registration and all our problems went away. Before that, the bt102's would randomly lose connection with asterisk (V1.2 BTW) Paul -- Paul Hewlett Technical Director Global Call Center Solutions Ltd, 2nd Floor, Milnerton Mall Cnr Loxton & Koeberg Roads, 7435 Milnerton www.gccs.co.za Tel: +27 86 111 3433 Fax: +27 86 111 3520 Cel: +27 76 072 7906 VOIP: 087 750 7260 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
