I'd be willing to bet you have "r" in your dialout string (i.e.
something like: Dial(${TRUNK}/${EXTEN},120,r)...Get rid of that in the outbound dialing, and you otta be ok. Jeremy Jones > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of > Bruce Komito > Sent: Friday, June 25, 2004 8:52 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] forced ring on dial? > > I am routing outgoing calls through a sip gateway. The calls > go through > no problem, however the ringing in the callers ear begins as > soon as the > last digit is dialed. This has two nasty side effects. > First, the caller > hears 1-2 more rings than the callee. Second, and more > importantly, if > the callee's line is busy, the caller continues to get hear > ringing, even > though the gateway has returned a busy indication. > > The whole problem seems to be * is not waiting for the proper call > progress signal from the sip gateway before giving the caller a ring > indication. Is there any way to control this so that * waits for call > progress from the gateway before giving the caller the appropriate > indication, i.e., ring or busy tone? I have been told this > is a result of > setting * to "forced ring" and this should be turned off, but > of course, > on * it is probably called something else. > > Thanks > > Bruce Komito > High Sierra Networks, Inc. > www.servers-r-us.com > (775) 236-5815 > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
