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Paging Problem

Posted by mike82 on Fri, 10/08/2010
Forums
Thirdlane platform General Questions

When I try to do paging I get a message sayign not a valid conference. It plays on all the phones I selected for the feature code. I know the script creates a conference room to do the page but how do i actually get it to work?

tl-page

exten => s,1,Set(CHANS=${ARG1})
exten => s,n,Set(ORIGCHANS=${CHANS}) ; Keep a copy
exten => s,n(loopstart),While($[${LEN(${CHANS})} > 0])
exten => s,n,Set(TESTCHAN=${CUT(CHANS,&,1)})

; Reset CHANS to the substring of CHANS that removes the length of the TESTCHAN variable, plus 1 additional character (the '&')

exten => s,n,Set(CHANS=${CHANS:${MATH(${LEN(${TESTCHAN})}+1)}})
exten => s,n,ChanIsAvail(${TESTCHAN},s)
exten => s,n,NoOp(TESTCHAN is "${TESTCHAN}" and and AVAILSTATUS is "${AVAILSTATUS}")
exten => s,n,GotoIf($[${AVAILSTATUS} = 0]?chanisavail) ; availstatus of 0 means channel is available
exten => s,n,Goto(loopstart) ; Loop back to top, don't add TESTCHAN to AVAILCHANS
exten => s,n(chanisavail),Set(AVAILCHANS=${AVAILCHANS}&${TESTCHAN})
exten => s,n,EndWhile
exten => s,n,Set(AVAILCHANS=${AVAILCHANS:1}) ; remove the leading '&'
exten => s,n,NoOp(Of original channels "${ORIGCHANS}", the available ones are "${AVAILCHANS}")
exten => s,n,SIPAddHeader(Alert-Info: )
exten => s,n,Set(_SIP_URI_OPTIONS=intercom=true)
exten => s,n,SIPAddHeader(Call-Info: <>\;answer-after=0)
exten => s,n,Page(${AVAILCHANS})

Submitted by mike82 on Mon, 10/18/2010 Permalink

If you mean the SIP extensions that get the page then yes. If you mean something else then could you explain please. It use to work so I'm wondering what happened.

Submitted by mike82 on Mon, 10/18/2010 Permalink

The first arguement is "phone to ring" with the option "phone(s)/line(s)" selected.
When I create a feature code i'm given the menu box for sip/ext-tenant name ext. I select a couple and the same "invalid conference" message plays on all the extensions I selected.

Does the script need to be tweaked? That's the original unless someone muddled arround with it.

Submitted by mike82 on Mon, 10/18/2010 Permalink

Yes, its calling each ext. I see each one ring in the log followed by the above message on each. And hear them on the phones near me.

In the log i get the following:

answered

warning 25205:app_meetme.c:800 build_conf:unable to acces psuedo device

playing 'conf-invalid'

It's on all the extension being paged.

Submitted by mike82 on Tue, 10/19/2010 Permalink

So because i'm using a SIP trunk to a gateway and no longer have my Digium T1 card in my PBX I can't create a conference? Could i reload the DAHDI drivers without a card and not have it cause problems?

Submitted by eeman on Tue, 10/19/2010 Permalink

have you tried installing recent dahdi drivers and then a make distclean and make install of your asterisk source code to use the dummy timing source?