Callfiles to Meetme Fails (was: RE: [asterisk-users] Using Local Channels with Originate)

Matthew Rubenstein email at mattruby.com
Mon Feb 5 17:21:07 MST 2007


	I have Meetme conferencing compiled for Debian as per
http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the
outgoing dir, and it intitiates a call to a local extension as a
channel, but the call seems to block on the Meetme() command. That
extension completes the outgoing Dial(SIP) command to my phone,
announcing that leg is the only member of the conference, and just
waits. If I hang up, the exit status is reported by Asterisk (in logfile
and CLI), then Asterisk jumps to the callfile's extension, which
completes the outgoing Dial(SIP) to the other phone, which then gets the
announcement that it is the only member of that conference. Why does it
block, instead of proceeding to the second leg and conferencing it in?

meetme.conf :---------------------------------------------
[rooms]
conf => 1234
EOF--------------------------------------------------------------------

from extensions.conf :-------------------------------------------------
[ext-out]
  exten => callFrom,1,Noop(Calling SIP/${callFrom}@${sipCarrier})
  exten => callFrom,n,Dial(SIP/${callFrom}@${sipCarrier},45,M(conf-from^
${callTo})g)
  exten => callFrom,n,Noop(Done dialing from)

  exten => callTo,1,Noop(Calling SIP/${callTo}@${sipCarrier})
  exten => callTo,n,Dial(SIP/${callTo}@${sipCarrier},45,M(conf-to^999)g)
  exten => callTo,n,Noop(Done dialing to)


[macro-conf-from]
        ; ARG1: callTo
        exten => s,1,Noop(in macro-conf-from)
        exten => s,n,Noop(before MeetMe: ${ARG1})
        exten => s,n,MeetMe(1234)
        exten => s,n,Noop(after MeetMe: ${ARG1})
EOF--------------------------------------------------------------------

out.call :-------------------------------------------------------------
Channel: Local/callFrom at ext-out/n

Callerid: 12126661234

Context: ext-out
Extension: callTo
Priority: 1

Set: callFrom=12126661234
Set: callTo=12127779999
Set: callerID=12126661234
Set: sipCarrier=sipcarrier
EOF--------------------------------------------------------------------



On Mon, 2007-02-05 at 15:52 -0700,
asterisk-users-request at lists.digium.com wrote:
> Date: Mon, 05 Feb 2007 17:37:31 -0500
> From: David Boyd <dboyd at ignitetrx.com>
> Subject: RE: [asterisk-users] Using Local Channels with Originate
> To: Asterisk Users Mailing List - Non-Commercial Discussion
>         <asterisk-users at lists.digium.com>
> Message-ID: <1170715051.14822.2.camel at dbdell.fullmoonsoft.com>
> Content-Type: text/plain; charset=utf-8
> 
> On Mon, 2007-02-05 at 13:21 -0800, Michael Collins wrote:
> > I havent quite gotten this working yet but I am going to update the
> > thread with what I have learned. Maybe this will help the next guy
> who
> > tries to figure this out
> > 
> >  
> > 
> > The trick to using the DIALSTATUS seems to be to put it in the
> handler
> > for the h (hang-up extension). 
> > 
> >  
> > 
> >     [outdialer]
> > 
> >     exten => 100, 1, Dial(${numberToDial})
> > 
> >     exten => h, 1, Goto(s-${DIALSTATUS},1)
> > 
> >  
> > 
> >     exten => s-ANSWER,1,NoOp("Answered")
> > 
> >     exten => s-BUSY,1,NoOp("Busy")
> > 
> >     exten => s-NOANSWER,1,NoOp("Not answered")
> > 
> >     exten => s-CANCEL,1,NoOp("Cancelled")
> > 
> >     exten => s-CONGESTION,1,NoOp("Fast busy")
> > 
> >     exten => s-CHANUNAVAIL,1,NoOp("Channel unavailable")
> > 
> >  
> > 
> >     [dialerplan]
> > 
> >     exten => s,1,Background(demo-congrats)
> > 
> >     exten => s,n,WaitExten
> > 
> >     so on ...
> > 
> >  
> > 
> > Here are the manager commands I am using:
> > 
> >  
> > 
> >     Action: login
> > 
> >     Username: test
> > 
> >     Secret: nottelling
> > 
> >  
> > 
> >     Action: originate
> > 
> >     Channel: Local/100 at outdialer/n
> > 
> >     Context: dialerplan
> > 
> >     Extension: s
> > 
> >     Priority: 1
> > 
> >     Variable: numberToDial=ZAP/4/1234567890
> > 
> >  
> > 
> >     Action: logoff
> > 
> >  
> > 
> > I am always getting ANSWERED for ${DIALSTAUS} so something is not
> > quite right. Hopefully I am getting closer.
> > 
> >  
> > 
> >  
> > 
> > Brian,
> > 
> >  
> > 
> > What kind of Zap hardware/telco lines are you using?  I am using PRI
> > and I am able to get a dial status in the hangup extension.  The
> > problem I run into is that I get NO ANSWER as the hangup cause even
> > for invalid phone numbers I also get cluttered CDRs.  In the
> > meantime Im working on a solution that I hope will give the best of
> > both worlds.  Im relying on the API events instead of local
> channels.
> > Ill post more information when Ive made more progress.  However,
> > Ive made 2500 test calls and I havent lost a single
> > OriginateSuccess or OriginateFailure event.  (Im keying on these,
> > specifically the OriginateFailure event because it has a Reason
> > value that gets populated: 0=Invalid, 3=No Ans, 5=Busy.)
> > 
> >  
> > 
> > Hope to have more info posted this week.
> > 
> >  
> > 
> > -MC
> >  
-- 

(C) Matthew Rubenstein



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