Hello Phil, please check the following details in your asterisk configuration and on your phones. These are the settings that work for me:
sip.conf ------------ [general] limitonpeers=yes allowsubscribe=yes notifyringing=yes notifyhold=yes useclientcode=yes canreinvite=yes [user1] secret=user1 host=dynamic username=user1 callerid="user1 <97>" dtmfmode=rfc2833 context=local type=friend callgroup=1 pickupgroup=1 qualify=yes vmexten=80297 call-limit=20 subscribecontext=local extensions.conf -------------------- exten => 97,hint,SIP/user1 exten => 98,hint,SIP/smguenther On the SNOM phones: Support broken Registrar: ON Use user:phone: OFF Filter Packets from Registrar: OFF Function Key P6: ACTIVE / EXTENSION / <sip:[EMAIL PROTECTED]> Hope that helps, Stefan -- ******************************************** in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Geschaeftsfuehrer Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de ******************************************** Schulungen Installationen Beratung Support Voice-over-IP-Loesungen ******************************************** _______________________________________________ --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