i did and its not working here is console output. We have 8910-8920 meetme conf room. below i am dialing 8991 for test invalid and its not working..
Packet timed out after 32000ms with no response == Using SIP RTP CoS mark 5 -- Executing [7580@from-sip:1] Goto("SIP/7527-00000030", "ivr-meetme,s,1") in new stack -- Goto (ivr-meetme,s,1) -- Executing [s@ivr-meetme:1] Answer("SIP/7527-00000030", "") in new stack -- Executing [s@ivr-meetme:2] Wait("SIP/7527-00000030", "1") in new stack -- Executing [s@ivr-meetme:3] BackGround("SIP/7527-00000030", "conf-getconfno") in new stack -- <SIP/7527-00000030> Playing 'conf-getconfno.ulaw' (language 'en') -- Executing [s@ivr-meetme:4] WaitExten("SIP/7527-00000030", "20") in new stack == CDR updated on SIP/7527-00000030 -- Executing [8991@ivr-meetme:1] MeetMe("SIP/7527-00000030", "8991,cMp") in new stack == Parsing '/etc/asterisk/meetme.conf': == Found == Spawn extension (ivr-meetme, 8991, 1) exited non-zero on 'SIP/7527-00000030' shirley*CLI> > Date: Wed, 6 Apr 2011 14:37:20 -0700 > From: asterisk....@sedwards.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] asterisk meetme invalid extension > > On Wed, 6 Apr 2011, satish patel wrote: > > > I have following dialplan for meetme and i want if someone type wrong > > meetme extension it should say invalid extension. But look like > > following doesn't work. its just hangup if i type wrong number. how to > > fix this code.. > > > > exten => i,n,Playback(pbx-invalid) > > The priority should be 1. > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users