Don Fanning wrote:
Taking in everyone's suggestions (added a username line also) here is what I got. Still no joy ---*CLI> *CLI> *CLI> -- Executing SetCallerID("SIP/100-b225", ""xxxx"") in new stack -- Executing Dial("SIP/100-b225", "IAX2/[EMAIL PROTECTED]/0015163011118") in new stack -- Called [EMAIL PROTECTED]/0015163011118 -- Hungup 'IAX2/voipbuster/6' == No one is available to answer at this time -- Executing Congestion("SIP/100-b225", "") in new stack == Spawn extension (internalselections, 90015163011118, 3) exited non-zero on 'SIP/100-b225'
Put a Noop(HANGUPCAUSE=${HANGUPCAUSE}) and a Noop(DIALSTATUS=${DIALSTATUS}) as the two priorities after your Dial to see WHY the call was hungup.
-- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 r: Generate a ringing tone for the calling party, passing no audio from the called channel(s) until one answers. Use with care and don't insert this by default into all your dial statements as you are killing call progress information for the user. Really, you almost certainly do not want to use this. Asterisk will generate ring tones automatically where it is appropriate to do so. "r" makes it go the next step and additionally generate ring tones where it is probably not appropriate to do so. _______________________________________________ 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
