Asterisk
  1. Asterisk
  2. ASTERISK-5702

[patch] Alternative SIP call pickup with Caller ID displayed

    Details

    • Type: New Feature New Feature
    • Status: Closed
    • Severity: Major Major
    • Resolution: Won't Fix
    • Affects Version/s: None
    • Target Release Version/s: None
    • Component/s: None
    • Labels:
      None
    • Mantis ID:
      5853
    • Regression:
      No

      Description

      I changed the patch for this fit for 1.0.8 and 1.0.9 version of *.
      I am no more able to do this for the 1.2.0 version.

        Activity

        Hide
        Olle Johansson added a comment -

        Who can disclaim this code legally?

        Show
        Olle Johansson added a comment - Who can disclaim this code legally?
        Hide
        kib added a comment -

        I will send the disclaimer fax http://www.digium.com/disclaimer.txt to +1-256-971-6890.

        We paid Martin Pycko (m78pl@yahoo.com) for this patch.

        Show
        kib added a comment - I will send the disclaimer fax http://www.digium.com/disclaimer.txt to +1-256-971-6890. We paid Martin Pycko (m78pl@yahoo.com) for this patch.
        Hide
        Kevin P. Fleming (Inactive) added a comment -

        I don't understand this at all. There is no documentation on what this is or how it is useful, just that it is an 'alterantive' method. Why would someone want to use this? How is better/different? Without any documentation, we don't even know what it is supposed to do, let alone whether it actually does it or not.

        Show
        Kevin P. Fleming (Inactive) added a comment - I don't understand this at all. There is no documentation on what this is or how it is useful, just that it is an 'alterantive' method. Why would someone want to use this? How is better/different? Without any documentation, we don't even know what it is supposed to do, let alone whether it actually does it or not.
        Hide
        kib added a comment -

        This is from http://www.voip-info.org/wiki/view/Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP
        **********************
        Call Pickup with CID info. (SIP)

        When a call is transferred from a ringing SIP-Phone to an other SIP-Phone using the *8 feature no call info is displayed.
        This is offcourse logical because the set is dialing out to *8 and therefore no Call Info can be displayed.

        What we would like to see is a call-pickup feature which works like this:

        A call arrives at a SIP-phone (Extension "X").

        1. A user diales *8 on Extension "Z" (or any other featurecode that we would like)
        2. Asterisk immediatly ends the call, but remembers the Extension of the person who dialed *8 (Extension "Z").
        3. Asterisk might wanna wait for 1 second (to let Extension "Z" become idle)
        4. Asterisk then transfers the incoming call on Extension "X" to Extension "Z".

        This offcourse will result in a new incoming call on extension "Z" textwithtext CID Info.
        An extra advantage of this is that the person on extension "Z" can choose whether to answer the call or not depending on the CID information.
        **********************
        This is for example a standard feature of standard PBX system. This is very usefull.

        Show
        kib added a comment - This is from http://www.voip-info.org/wiki/view/Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP ********************** Call Pickup with CID info. (SIP) When a call is transferred from a ringing SIP-Phone to an other SIP-Phone using the *8 feature no call info is displayed. This is offcourse logical because the set is dialing out to *8 and therefore no Call Info can be displayed. What we would like to see is a call-pickup feature which works like this: A call arrives at a SIP-phone (Extension "X"). 1. A user diales *8 on Extension "Z" (or any other featurecode that we would like) 2. Asterisk immediatly ends the call, but remembers the Extension of the person who dialed *8 (Extension "Z"). 3. Asterisk might wanna wait for 1 second (to let Extension "Z" become idle) 4. Asterisk then transfers the incoming call on Extension "X" to Extension "Z". This offcourse will result in a new incoming call on extension "Z" textwithtext CID Info. An extra advantage of this is that the person on extension "Z" can choose whether to answer the call or not depending on the CID information. ********************** This is for example a standard feature of standard PBX system. This is very usefull.
        Hide
        Olle Johansson added a comment -

        Closing this report, leaving it accessible in the bug tracker for download. It will not be included in Asterisk due to architectural reasons. We are instead concentrating on fixing connected line ID signalling as being implemented in another patch.

        Thanks for your work and willingness to contribute to Asterisk!

        /O

        Show
        Olle Johansson added a comment - Closing this report, leaving it accessible in the bug tracker for download. It will not be included in Asterisk due to architectural reasons. We are instead concentrating on fixing connected line ID signalling as being implemented in another patch. Thanks for your work and willingness to contribute to Asterisk! /O

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            Dates

            • Created:
              Updated:
              Resolved:

              Development