Just check DND if it's on on the phone or not. What is the CLI output when you try making a phone call? Why don't you try it with a later version of astrisk and a Phone?
On Feb 13, 2008 10:58 AM, Giordano Grandis <[EMAIL PROTECTED]> wrote: > > > Hi all gusy, > i have a big problem with gxp2000 and asterisk 1.0.9 The phones after a few > go in "busy" state, if you call it get the busy tone but the phone can male > any type of call. > This is my sip.conf > > [502] > language = it > username = 502 > secret = <password> > host = dynamic > type = friend > context = local > canreinvite = yes > dtmfmode = info > callgroup = 1 > pickupgroup = 1 > callerid = 502 <502> > > Under Grandstream's support suggest, I set "Use randmom port" to yes and > "Nat traversal (STUN)" to "No, but send keep alive" but without success. > This is the firmware version: Program-- 1.1.5.15 Bootloader-- 1.1.5.6 > > Anyone can help me ? > > Thanks in advance > > Giordano > > > No virus found in this outgoing message. > Checked by AVG Free Edition. > Version: 7.5.516 / Virus Database: 269.20.4/1275 - Release Date: 12/02/2008 > 15.20 > > _______________________________________________ > -- 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 > _______________________________________________ -- 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