[Asterisk-Users] transfers with zap channel

Mon Jan 17 21:29:50 MST 2005

Ah, suddenly everything becomes clear.   I've never looked in features.conf before.  I now understand that 700 is supposed to intitiate the call park, and it's taking precidence over the extension I was trying to dial of 7007.  I've changed the call parking extension and now I can do regular attended and unattended transfers without having to park the call... 
(note to anyone else changing features.conf, you have to 'restart' asterisk, a 'reload' won't do).

thanks a bunch for the help, guys...

Paul
  ----- Original Message ----- 
  From: Lyle Giese 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Monday, January 17, 2005 8:20 PM
  Subject: Re: [Asterisk-Users] transfers with zap channel

  Have you looked at features.conf?

  Lyle
    ----- Original Message ----- 
    From: Paul Fielding 
    To: Asterisk Users Mailing List - Non-Commercial Discussion 
    Sent: Monday, January 17, 2005 8:53 PM
    Subject: Re: [Asterisk-Users] transfers with zap channel

    The outside line isn't actually being dropped - the outside line hanging up is me hanging up the outside line after finding that my transfer failed.

    I must be not understanding how the flash-hook works then.  My understanding was that when I flash-hook and get a second dialtone I should be able to dial the extention I want to reach (7007 is another extension, via SIP).  Normally, if I pick up the analog phone and dial 7007 it rings the extention fine.   Apparently, though, when you get that second dialtone, it has different rules?   I haven't been able to find documentation on this, can it be found anywhere?  For example, why does dialing 700 park the call?  I haven't found anything on this... *shrug*...

    Paul

      ----- Original Message ----- 
      From: Lyle Giese 
      To: Asterisk Users Mailing List - Non-Commercial Discussion 
      Sent: Monday, January 17, 2005 7:22 PM
      Subject: Re: [Asterisk-Users] transfers with zap channel

      How long between getting parked is the orginal call dropping?  

      Depending on your dialplan, yes dialing 700x will almost immediately send the call to call parking. (IMHO, poor extension planning can also cause this.)

      I don't use the t or T options<PERIOD>.  IMHO, you just lose the ability to use the # key and confused the heck out of my users.  Took it out and use the flash method only in my dial plan.  Dial 700, park the call.  Dial the other extension, tell them to pick up 701.  Or use meetme for conference calling?

      I know I need to play with three way calling here also.

      Lyle

        ----- Original Message ----- 
        From: Paul Fielding 
        To: Asterisk Users Mailing List - Non-Commercial Discussion 
        Sent: Monday, January 17, 2005 6:12 PM
        Subject: [Asterisk-Users] transfers with zap channel

        Ok, I've seen discussion before on doing transfers (attended and unattended), there seems to be much confusion over it.

        As things sit, I've been trying (unsuccessfully) to do transfers with a zap channel (analog phone attached to TDM400).  I have no idea what I'm missing.  My current understanding is that I need to have transfer=yes in zapata.conf, do a flash hook, dial the 2nd number, flash hook again and we're linked (attended).   Then if I hang up the call will be transfered.

        However, when I try to do this things don't work.   Here's what I do:

        - connection is made between Zap/3 (analog phone) and Zap/1 (outside line).
        - flash hook to get dialtone (I do get dialtone)
        - attempt to transfer to extension 7007 - I dial 7007
        - after dialing the 2nd zero, and before dialing the 2nd seven, I hear Asterisk announce (seven - zero - one) and then Zap/3 is hung up (I get a busy signal).  Zap/1 gets parked.

        Here's what the log shows:

            -- Zap/1-1 answered Zap/3-1
            -- Attempting native bridge of Zap/3-1 and Zap/1-1
            -- Started three way call on channel 1
            -- Started music on hold, class 'default', on Zap/3-1
            -- Attempting native bridge of Zap/3-1 and Zap/1-1
            -- Starting simple switch on 'Zap/1-2'
            -- Started music on hold, class 'default', on Zap/3-1
          == Parked Zap/3-1 on 701. Will timeout back to dostuff,7001,1 in 45 seconds
            -- Added extension '701' priority 1 to parkedcalls
            -- Playing 'digits/7' (language 'en')
            -- Hungup 'Zap/1-1'
          == Spawn extension (dostuff, 7001, 1) exited non-zero on 'Parked/Zap/3-1<ZOMBIE>'
            -- Stopped music on hold on Parked/Zap/3-1<ZOMBIE>
            -- Playing 'digits/0' (language 'en')
            -- Playing 'digits/1' (language 'en')
            -- Parking call to 'Zap/1-2'
            -- Hungup 'Zap/1-2'
            -- Stopped music on hold on Zap/3-1
          == Zap/3-1 got tired of being parked
            -- Hungup 'Zap/3-1'

        I'm not sure what I'm missing.  Apparently something to do with parked calls, so I must be misunderstanding how do to the call transfer.

        I've also tried enabling Asterisk transfer on the channel (exten => 7010,1,Dial(${CORDLESS},20,tT)).
        My understanding of this method is that this allows one to hit the pound (#) to start a transfer.  Yet pound does nothing.  Is it fair to assume that the tT only works on SIP channels, or am I missing something else.

        Any help is much appreciated....

        Paul

------------------------------------------------------------------------

        _______________________________________________
        Asterisk-Users mailing list
        Asterisk-Users at lists.digium.com
        http://lists.digium.com/mailman/listinfo/asterisk-users
        To UNSUBSCRIBE or update options visit:
           http://lists.digium.com/mailman/listinfo/asterisk-users

--------------------------------------------------------------------------

      _______________________________________________
      Asterisk-Users mailing list
      Asterisk-Users at lists.digium.com
      http://lists.digium.com/mailman/listinfo/asterisk-users
      To UNSUBSCRIBE or update options visit:
         http://lists.digium.com/mailman/listinfo/asterisk-users

----------------------------------------------------------------------------

    _______________________________________________
    Asterisk-Users mailing list
    Asterisk-Users at lists.digium.com
    http://lists.digium.com/mailman/listinfo/asterisk-users
    To UNSUBSCRIBE or update options visit:
       http://lists.digium.com/mailman/listinfo/asterisk-users

------------------------------------------------------------------------------

  _______________________________________________
  Asterisk-Users mailing list
  Asterisk-Users at lists.digium.com
  http://lists.digium.com/mailman/listinfo/asterisk-users
  To UNSUBSCRIBE or update options visit:
     http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050117/096b466a/attachment.htm