App Conference Dial Plan help

Asterisk Asterisk Support
1

Hello,

Finally got app conference working thanks to g2010 (thanks again)

Now i need to get my dialplan sorted to offer a “normal” level of funtionality.

Heres what Im looking to acheive:

  1. User dials in over SIP trunk (sipgate)
  2. Asked to enter a meeting code. (no pin)
  3. User goes into conference and is asked to record name
  4. User is annouced

I have been following this guide

voip-info.org/wiki/view/Aste … Conference

and trying to modify this code:

[code][conferences]
exten => s,1,Wait(1)
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,8
exten => s,4,BackGround(enter-conf-call-number)
exten => s,5,Waitexten(20)
exten => s,6,Goto(s,4)

exten => _XXXX,1,DBget(pass=conferences/${EXTEN}) ; implied jump to 101 if it doesn’t exist
exten => _XXXX,2,GotoIf($[“xxx${pass}” = “xxxNONE”]?30) ; if no pass, jump directly to conf
exten => _XXXX,3,Read(secret,pls-enter-conf-password,10)
exten => _XXXX,4,GotoIf($["${pass}" != “${secret}”]?40:30)

exten => _XXXX,30,Conference(${EXTEN}/MTV)
exten => _XXXX,31,Hanugup

exten => _XXXX,40,Playback(conf-invalidpin)
exten => _XXXX,41,Hangup

exten => _XXXX,101,Playback(conf-invalid)
exten => _XXXX,102,Hangup [/code]

However it seems Im missing something or something here as I know the above is incomplete for my scenario, but not sure what Im missing.

Here is my extensions.conf as it stands:

Trying to get ext 2000 to drop users into conference.

[code][default]
exten=>1000,1,Dial(SIP/1000,30)
exten=>1000,2,Voicemail(u1000)
exten=>2000,1,Answer(2)
exten=>2000,2,goto(conferences,s,1)

[conferences]
exten => s,1,Wait(1)
exten => s,2,DigitTimeout,5
exten => s,3,ResponseTimeout,8
exten => s,4,BackGround(enter-conf-call-number)
exten => s,5,Waitexten(20)
exten => s,6,Goto(s,4)

exten => _XXXX,1,database get(pass=conferences/${EXTEN}) ; implied jump to 101 if it doesn’t exist
exten => _XXXX,2,GotoIf($[“xxx${pass}” = “xxxNONE”]?30) ; if no pass, jump directly to conf
exten => _XXXX,3,Read(secret,pls-enter-conf-password,10)
exten => _XXXX,4,GotoIf($["${pass}" != “${secret}”]?40:30)

exten => _XXXX,30,Conference(${EXTEN}/MTV)
exten => _XXXX,31,Hanugup

exten => _XXXX,40,Playback(conf-invalidpin)
exten => _XXXX,41,Hangup

exten => _XXXX,101,Playback(conf-invalid)
exten => _XXXX,102,Hangup

[from_sipgate]
exten => 1046740,1,Answer
exten => 1046740,2,Dial(SIP/1000,30)
[/code]

ERROR is:

-- Executing [s@conferences:1] Wait("SIP/1000-009d11a0", "1") in new stack [Sep 21 08:41:20] WARNING[6219]: pbx.c:1779 pbx_extension_helper: No application 'DigitTimeout' for extension (conferences, s, 2)

So the questions are:

Is an application missing? (its v1.4 by the way as needed for appconference)

How do I get the incoming callers into conference. I guess I need to forward them to an extension that sends them to conference? Or is it context?

How do I get the announcements to work etc?

Thanks in advance.

2

Not having dahdi-dummy will also cause you problems if your phones cannot have silence suppression disabled.

3

Yep, its looking like this particular cloud is out. Calls are incredibly noisy.

4

i have a version of app_conference which works great on asterisk 1.4. I have used it on 1.4.21, 1.4.25, and now 1.4.26. I can confirm it works great with ulaw and g729, can’t vouch for anything else.

I have a stripped down version which I use for our VACD platform, and one with some announcement features which I use for our office PBX.

If you are interested, send me a PM.

5

thanks mate all sorted on the install now, just need to get my head around a dial plan

6

Right, sorted a very rudimentary dial plan.

[code]; This is the extension that people actually dial to get the bridge
exten => 99,1,Ringing()
exten => 99,2,wait(5)
exten => 99,3,goto(conferences,s,1)

[conferences]
exten => s,1,Wait(1)
;exten => s,2,DigitTimeout,5
;exten => s,3,ResponseTimeout,8
exten => s,2,BackGround(conf-getconfno)
exten => s,3,Waitexten(20)
exten => s,4,Goto(s,2)

exten => 400152#,1,Playback(auth-thankyou)
exten => 400152#,2,Playback(conf-placeintoconf)
exten => 400152#,3,Wait(2)
exten => 400152#,4,Conference(${EXTEN}/Smn)

exten => 500199#,1,Playback(auth-thankyou)
exten => 500199#,2,Playback(conf-placeintoconf)
exten => 500199#,3,Wait(2)
exten => 500199#,4,Conference(${EXTEN}/Smn)

exten => i,1,Playback(conf-invalid)
exten => i,2,goto(s,2)

;exten => _XXXX,101,Playback(conf-invalid)
;exten => _XXXX,102,goto(s,2)

[from_sipgate]
exten => 1046740,1,Ringing()
exten => 1046740,2,Wait(5)
exten => 1046740,3,goto(conferences,s,1)[/code]

As i will only be issuing known pin codes this works fine.

Things to fix

  1. Music on hold doesnt work even when passing the “m” trigger. * complains of unknown command

Sep 21 13:54:32] WARNING[9329]: member.c:1170 create_member: received invalid flag, chan => SIP/1000-009ee800, flag => m -- <SIP/1000-009ee800> Playing 'conf-onlyperson' (language 'en')

  1. Would be nice to have a prompt for callers name, record, and then playback once in call but once you have invoked “conference” you no option to. hmmmm.