MeetMe Conference Invalid PIN

I have been having problems with setting up a conference call. I getting getting a message “That PIN is invalid for the conference”. I have verified that the correct PIN (1234 in this case) is being passes correctly to the system. The only fix I could find through other forums said to chown -r asterisk:asterisk /dev/zap/ . /dev/dahdi/ is already showing asterisk:asterisk.

Any ideas?

Asterisk 1.4.33.1
FreePBX 2.8.0

meetme_additional.conf

conf => 99900,1234

-- Executing [[email protected]_OKC_SIP:1] Macro("SIP/2049-0000001c", "user-callerid|") in new stack -- Executing [[email protected]:1] Set("SIP/2049-0000001c", "AMPUSER=2049") in new stack -- Executing [[email protected]:2] GotoIf("SIP/2049-0000001c", "0?report") in new stack -- Executing [[email protected]:3] ExecIf("SIP/2049-0000001c", "1|Set|REALCALLERIDNUM=2049") in new st ack -- Executing [[email protected]:4] Set("SIP/2049-0000001c", "AMPUSER=2049") in new stack -- Executing [[email protected]:5] Set("SIP/2049-0000001c", "AMPUSERCIDNAME=CONF - Main") in new stack -- Executing [[email protected]:6] GotoIf("SIP/2049-0000001c", "0?report") in new stack -- Executing [[email protected]:7] Set("SIP/2049-0000001c", "AMPUSERCID=2049") in new stack -- Executing [[email protected]:8] Set("SIP/2049-0000001c", "CALLERID(all)="CONF - Main" <2049>") in n ew stack -- Executing [[email protected]:9] ExecIf("SIP/2049-0000001c", "0|Set|CHANNEL(language)=") in new stac k -- Executing [[email protected]:10] GotoIf("SIP/2049-0000001c", "0?continue") in new stack -- Executing [[email protected]:11] Set("SIP/2049-0000001c", "__TTL=64") in new stack -- Executing [[email protected]:12] GotoIf("SIP/2049-0000001c", "1?continue") in new stack -- Goto (macro-user-callerid,s,19) -- Executing [[email protected]:19] NoOp("SIP/2049-0000001c", "Using CallerID "CONF - Main" <2049>") i n new stack -- Executing [[email protected]_OKC_SIP:2] Set("SIP/2049-0000001c", "MEETME_ROOMNUM=99900") in new stack -- Executing [[email protected]_OKC_SIP:3] Set("SIP/2049-0000001c", "MAX_PARTICIPANTS=0") in new stack -- Executing [[email protected]_OKC_SIP:4] Set("SIP/2049-0000001c", "MEETME_MUSIC=") in new stack -- Executing [[email protected]_OKC_SIP:5] GotoIf("SIP/2049-0000001c", "0?READPIN") in new stack -- Executing [[email protected]_OKC_SIP:6] Answer("SIP/2049-0000001c", "") in new stack -- Executing [[email protected]_OKC_SIP:7] Wait("SIP/2049-0000001c", "1") in new stack -- Executing [[email protected]_OKC_SIP:8] Set("SIP/2049-0000001c", "PINCOUNT=0") in new stack -- Executing [[email protected]_OKC_SIP:9] Read("SIP/2049-0000001c", "PIN|enter-conf-pin-number||||") in new stack -- <SIP/2049-0000001c> Playing 'enter-conf-pin-number' (language 'en') -- User entered '1234' -- Executing [[email protected]_OKC_SIP:10] GotoIf("SIP/2049-0000001c", "1?USER") in new stack -- Goto (CC_OKC_SIP,99900,18) -- Executing [[email protected]_OKC_SIP:18] Set("SIP/2049-0000001c", "MEETME_OPTS=c") in new stack -- Executing [[email protected]_OKC_SIP:19] Goto("SIP/2049-0000001c", "STARTMEETME|1") in new stack -- Goto (CC_OKC_SIP,STARTMEETME,1) -- Executing [[email protected]_OKC_SIP:1] ExecIf("SIP/2049-0000001c", "0|SetMusicOnHold|") in new stack -- Executing [[email protected]_OKC_SIP:2] Set("SIP/2049-0000001c", "GROUP(meetme)=99900") in new stack -- Executing [[email protected]_OKC_SIP:3] GotoIf("SIP/2049-0000001c", "0?MEETMEFULL|1") in new stack -- Executing [[email protected]_OKC_SIP:4] MeetMe("SIP/2049-0000001c", "99900|c|1234") in new stack -- <SIP/2049-0000001c> Playing 'conf-invalidpin' (language 'en') == Spawn extension (CC_OKC_SIP, STARTMEETME, 4) exited non-zero on 'SIP/2049-0000001c' -- Executing [[email protected]_OKC_SIP:1] Macro("SIP/2049-0000001c", "hangupcall|") in new stack -- Executing [[email protected]:1] GotoIf("SIP/2049-0000001c", "1?skiprg") in new stack -- Goto (macro-hangupcall,s,4) -- Executing [[email protected]:4] GotoIf("SIP/2049-0000001c", "1?skipblkvm") in new stack -- Goto (macro-hangupcall,s,7) -- Executing [[email protected]:7] GotoIf("SIP/2049-0000001c", "1?theend") in new stack -- Goto (macro-hangupcall,s,9) -- Executing [[email protected]:9] Hangup("SIP/2049-0000001c", "") in new stack == Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/2049-0000001c' in macro 'hangupcall' == Spawn extension (CC_OKC_SIP, h, 1) exited non-zero on 'SIP/2049-0000001c'

get AsteriskNOW immediately,
it is available on asterisk’s website, simply install it and access FreePBX
then download the CONFERENCE module from
http://mirror.freepbx.org/modules/release/2.7/

and your work will become so easy
In 3 minutes you will have created your Conference room and it will work perfectly.

AsteriskNOW is introduced for the ease of work, so that people dont have to work manually in files o Asterisk server anymore.

With AsteiskNOW, ALL YOUR WORK IS ON GUI AND CONFERENCE ROOMS ARE NO BIG ISSUE

Is dahdi started? Meetme requires a timing source, run /etc/init.d/dahdi status and see if you get a line like:

### Span  1: DAHDI_DUMMY/1 "DAHDI_DUMMY/1 (source: RTC) 1" (MASTER)

Without a timing source meetme is terminated:

== Spawn extension (CC_OKC_SIP, STARTMEETME, 4) exited non-zero on 'SIP/2049-0000001c'

I’m having same problem. I’ve restarted dahdi.
The conference pin will only valid when I call from softphone, but invalid when I call from hard phone.
What’s the difference between calling from softphone to hardphone?