freePBX conference

hi…

iam very new to freepbx. i want to create conferences in freepbx.
i hv installed freepbx2.8.0.1 .and created conference .but while i am dialing that conference number from my sip extension.it was saying invalid pin and hangup the call…
cli mode displays like this:

Executing [[email protected]:1] Macro(“SIP/201-0000000b”, “user-callerid,”) in new stack
– Executing [[email protected]:1] Set(“SIP/201-0000000b”, “AMPUSER=201”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/201-0000000b”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/201-0000000b”, “1?Set(REALCALLERIDNUM=201)”) in new stack
– Executing [[email protected]:4] Set(“SIP/201-0000000b”, “AMPUSER=201”) in new stack
– Executing [[email protected]:5] Set(“SIP/201-0000000b”, “AMPUSERCIDNAME=sip201”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/201-0000000b”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/201-0000000b”, “AMPUSERCID=201”) in new stack
– Executing [[email protected]:8] Set(“SIP/201-0000000b”, “CALLERID(all)=“sip201” <201>”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/201-0000000b”, “0?continue”) in new stack
– Executing [[email protected]:10] Set(“SIP/201-0000000b”, “__TTL=64”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/201-0000000b”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] NoOp(“SIP/201-0000000b”, “Using CallerID “sip201” <201>”) in new stack
– Executing [[email protected]:2] Set(“SIP/201-0000000b”, “MEETME_ROOMNUM=9999”) in new stack
– Executing [[email protected]:3] Set(“SIP/201-0000000b”, “MAX_PARTICIPANTS=9”) in new stack
– Executing [[email protected]:4] Set(“SIP/201-0000000b”, “MEETME_MUSIC=”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/201-0000000b”, “0?READPIN”) in new stack
– Executing [[email protected]:6] Answer(“SIP/201-0000000b”, “”) in new stack
– Executing [[email protected]:7] Wait(“SIP/201-0000000b”, “1”) in new stack
– Executing [[email protected]:8] Set(“SIP/201-0000000b”, “PINCOUNT=0”) in new stack
– Executing [[email protected]:9] Read(“SIP/201-0000000b”, “PIN,enter-conf-pin-number,”) in new stack
– User disconnected

help me plz

You need to insall DAHDI or a valid timing source.

i’m already having dahdi linux complete package

Check that file ‘enter-conf-pin-number’ exists in /var/lib/astrisk/sounds/ in any format. Check that dahdi is loaded in Asterisk with command ‘dahdi show status’.

actually we didn’t have the dahdi card in my server .we are making the calls through sip only.when i dial conference no from sip extension it is saying invalid pin

‘enter-conf-pin-number’ this file is not exist in my directory /var/lib/astrisk/sounds/ .so how can we put that file in that directory .

plz help me

but when i dial the conference number it is asking pin .then i entered pin after that call was hangup

Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] NoOp(“SIP/201-00000249”, “Using CallerID “sip201” <201>”) in new stack
– Executing [[email protected]:2] Set(“SIP/201-00000249”, “MEETME_ROOMNUM=9999”) in new stack
– Executing [[email protected]:3] Set(“SIP/201-00000249”, “MAX_PARTICIPANTS=10”) in new stack
– Executing [[email protected]:4] Set(“SIP/201-00000249”, “MEETME_MUSIC=”) in new stack
– Executing [[email protected]:5] Set(“SIP/201-00000249”, “MEETME_RECORDINGFILE=/var/spool/asterisk/monitor/meetme-conf-rec-9999-1311408022.585”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/201-00000249”, “0?READPIN”) in new stack
– Executing [[email protected]:7] Answer(“SIP/201-00000249”, “”) in new stack
– Executing [[email protected]:8] Wait(“SIP/201-00000249”, “1”) in new stack
– Executing [[email protected]:9] Set(“SIP/201-00000249”, “PINCOUNT=0”) in new stack
– Executing [[email protected]:10] Read(“SIP/201-00000249”, “PIN,enter-conf-pin-number,”) in new stack
– <SIP/201-00000249> Playing ‘enter-conf-pin-number.gsm’ (language ‘en’)
– User entered ‘111’
– Executing [[email protected]:11] GotoIf(“SIP/201-00000249”, “1?USER”) in new stack
– Goto (ext-meetme,9999,16)
– Executing [[email protected]:16] Set(“SIP/201-00000249”, “MEETME_OPTS=r”) in new stack
– Executing [[email protected]:17] Goto(“SIP/201-00000249”, “STARTMEETME,1”) in new stack
– Goto (ext-meetme,STARTMEETME,1)
– Executing [[email protected]:1] ExecIf(“SIP/201-00000249”, “0?Set(CHANNEL(musicclass)=)”) in new stack
– Executing [[email protected]:2] Set(“SIP/201-00000249”, “GROUP(meetme)=9999”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/201-00000249”, “0?MEETMEFULL,1”) in new stack
– Executing [[email protected]:4] MeetMe(“SIP/201-00000249”, “9999,r,111”) in new stack
== Parsing ‘/etc/asterisk/meetme.conf’: == Found
== Parsing ‘/etc/asterisk/meetme_additional.conf’: == Found
== Spawn extension (ext-meetme, STARTMEETME, 4) exited non-zero on ‘SIP/201-00000249’
– Executing [[email protected]:1] Hangup(“SIP/201-00000249”, “”) in new stack
== Spawn extension (ext-meetme, h, 1) exited non-zero on ‘SIP/201-00000249’

Please, when I tell you to check if you have DAHDI installed, then check that you actually have it installed!
Meetme must have a timing source, ie, DAHDI to work. It does not matter if you only use SIP.

We have repeatedly told you that DAHDI is required for conferences. Even without a conference card you must have the Dahdi/Dummy driver running.