(SOLVED!)Conference room asks for pin twice

Hi all,

I have an issue when callers enter a conference room it asks for pin then waits and asks again. It says “please enter conference pin number”, then when you enter it it says it requests the pin again then after its entered for the second time it will allow you to enter. I would like to change it so that it only asks once and the user is only required to enter the pin once.

TIA

So you’re saying you identified an issue in app_meetme or app_confbridge, or you’re using someone’s packaged distribution? What does the dialplan look like?

Sorry, I am still somewhat new to asterisk… The conference rooms are setup with meetme.
This is the contents of meetme_additional.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;

conf => 5001,147,72113
conf => 5002,
conf => 5555,2322,72113

Forgive me, I tried to find the file with the dial plan but was unsuccessful, could you point me to which file that is contained in?

I can’t, sorry, FreePBX creates “exciting” dialplans.

If you can narrow the problem and determine that it’s app_meetme that’s not reading the PIN properly the first time, we can look into it.

Here is a snippet of the output with verbosity set to 3
Hope this helps

 -- Executing [5555@from-internal:9] Set("SIP/618-00000350", "PINCOUNT=0") in new stack
-- Executing [5555@from-internal:10] Read("SIP/618-00000350", "PIN,enter-conf-pin-number,,,,") in new stack
-- <SIP/618-00000350> Playing 'enter-conf-pin-number.gsm' (language 'en')
-- User entered '2322'
-- Executing [5555@from-internal:11] GotoIf("SIP/618-00000350", "1?USER") in new stack
-- Goto (from-internal,5555,19)
-- Executing [5555@from-internal:19] Set("SIP/618-00000350", "MEETME_OPTS=cIMs") in new stack
-- Executing [5555@from-internal:20] Goto("SIP/618-00000350", "STARTMEETME,1") in new stack
-- Goto (from-internal,STARTMEETME,1)
-- Executing [STARTMEETME@from-internal:1] ExecIf("SIP/618-00000350", "1?Set(CHANNEL(musicclass)=default)") in new stack
-- Executing [STARTMEETME@from-internal:2] Set("SIP/618-00000350", "GROUP(meetme)=5555") in new stack
-- Executing [STARTMEETME@from-internal:3] GotoIf("SIP/618-00000350", "0?MEETMEFULL,1") in new stack
-- Executing [STARTMEETME@from-internal:4] MeetMe("SIP/618-00000350", "5555,cIMs,") in new stack

== Parsing ‘/etc/asterisk/meetme.conf’: == Found
== Parsing ‘/etc/asterisk/meetme_additional.conf’: == Found
– Created MeetMe conference 1023 for conference ‘5555’
– <SIP/618-00000350> Playing ‘conf-getpin.gsm’ (language ‘en’)
– <SIP/618-00000350> Playing ‘vm-rec-name.gsm’ (language ‘en’)
– <SIP/618-00000350> Playing ‘beep.gsm’ (language ‘en’)
– x=0, open writing: /var/spool/asterisk/meetme/meetme-username-5555-1 format: sln, 0x1a754a68
– User ended message by pressing #
– <SIP/618-00000350> Playing ‘auth-thankyou.gsm’ (language ‘en’)
– <SIP/618-00000350> Playing ‘conf-onlyperson.gsm’ (language ‘en’)
– Started music on hold, class ‘default’, on SIP/618-00000350
– Stopped music on hold on SIP/618-00000350
– Started music on hold, class ‘default’, on SIP/618-00000350
– Stopped music on hold on SIP/618-00000350

When it plays ‘conf-getpin.gsm’ its the same message as 'enter-conf-pin-number.gsm’
It doesn’t show it in this log for some reason, but I have to enter the pin after it plays ‘conf-getpin.gsm’

IF there is an easy fix out there someone may post it to answer someones question in the future. I am no longer pursuing the answer to this problem because it appears to be a bug and is corrected in a newer release and I am currently building a new server with the new version that will correct this. Thanks for the time you spent thinking about this. I am going to mark as solved because an update will fix this issue for anyone that comes across this post in the future.

My Version:

  • Running Asterisk Version : Asterisk 1.8.5.0
  • Asterisk Source Version : 1.6.1.10
  • Dahdi Source Version : 2.2.0.2+2.2.0
  • Libpri Source Version : 1.4.10.2
  • Addons Source Version : 1.6.1.1