[Asterisk-Users] Call Transfer Problem

[Surajee Ratnayake]

yes, u are quite right, you can find this feature in almost every pbx now.

We are also wondering whether, presently some one is implementing this feature or not, if no body is doing that, we can
start on that

Surajee


  ----- Original Message ----- 
  From: George Lin 
  To: surajee at infotechs.lk 
  Sent: Wednesday, June 04, 2003 3:36 AM
  Subject: RE: [Asterisk-Users] Call Transfer Problem


  so, What should the call initiator do if s/he wants to transfer the call initiated by himself/herself, by using flash keypad or what else ?

  I can see such application can be used in some big office, where the BOSS always asks the secretary to make the call, once the call is connected, then the secretary can trasfer the call to the BOSS. in order to let the BOSS talk on the phone. am I right ?? 

  Please let me know once the feature is implemented.

  George Lin
    -----Original Message-----
    From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Surajee Ratnayake
    Sent: Monday, June 02, 2003 1:05 AM
    To: asterisk-users at lists.digium.com
    Subject: Re: [Asterisk-Users] Call Transfer Problem


    U get the following output when u execute the "show application Dial" command in the Asterisk prompt,


      -= Info about application 'Dial' =- 

    [Synopsis]:
      Place an call and connect to the current channel

    [Description]:
      Dial(Technology/resource[&Technology2/resource2...][|timeout][|options][|URL]):
    Requests  one  or more channels and places specified outgoing calls on them.
    As soon as a  channel  answers, the  Dial  app  will  answer the originating
    channel (if it needs to be answered) and will bridge a call with the channel
    which first answered. All other calls placed by the Dial app will be hunp up
    f a timeout is not specified, the Dial  application  will wait indefinitely
    until either one of the  called channels  answers, the user hangs up, or all
    channels return busy or  error. In general,  the dialler will return 0 if it
    was  unable  to  place  the  call, or the timeout expired.  However, if  all
    channels were busy, and there exists an extension with priority n+101 (where
    n is the priority of  the  dialler  instance), then  it  will  be  the  next
    executed extension (this allows you to setup different behavior on busy from
    no-answer).
      This application returns -1 if the originating channel hangs up, or if the
    call is bridged and  either of the parties in the bridge terminate the call.
    The option string may contain zero or more of the following characters:
          't' -- allow the called user transfer the calling user
          'T' -- to allow the calling user to transfer the call.
          'r' -- indicate ringing to the calling party, pass no audio until answered.
          'm' -- provide hold music to the calling party until answered.
          'd' -- data-quality (modem) call (minimum delay).
          'c' -- clear-channel data call (PRI-PRI only).
          'H' -- allow caller to hang up by hitting *.
          'C' -- reset call detail record for this call.
          'P[(x)]' -- privacy mode, using 'x' as database if provided.
      In addition to transferring the call, a call may be parked and then picked
    up by another user.
      The optionnal URL will be sent to the called party if the channel supports
    it.



    Surajee


      ----- Original Message ----- 
      From: George Lin 
      To: surajee at infotechs.lk 
      Sent: Monday, June 02, 2003 1:11 PM
      Subject: FW: [Asterisk-Users] Call Transfer Problem


      Hi,

       

      Which document  describes the Dial with “T” option ? Could you let me know or email it to me.

       

      Thanks,

       

      George Lin

       

      -----Original Message-----
      From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Surajee Ratnayake
      Sent: Sunday, June 01, 2003 9:10 PM
      To: asterisk-users at lists.digium.com
      Subject: [Asterisk-Users] Call Transfer Problem

       

       

      hi All,

       

      We are working on Soft-PBX using Asterisk.  This relates to CALL TRANSFERRING aspects of Asterisk.

       

      We were able to do one type of call transfering, ie, the called person can transfer the original call to another person.

       

      but we were unable to do the other, that is, call initiator him/her self couldn't transfer the call. Eventhough the documentation for Dial application intructs to use "T" to achieve that.

      and we learnt that it has not been implemented yet in Asterisk. Is this true? 

      Is some one workin on this issue? if the answer is NO, we can give a try to implement it, with a help of u all , ofcourse :-)

       (cos, we are quite new to asterisk-only 1 month of experience, but amazed of its great performance)

       

      Thank you very much,

       

      Surajee

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