Summary:ASTERISK-05702: [patch] Alternative SIP call pickup with Caller ID displayed
Reporter:kib (kibeki)Labels:
Date Opened:2005-11-25 07:46:45.000-0600Date Closed:2006-03-29 19:50:41.000-0600
Versions:Frequency of
Environment:Attachments:( 0) alternative_sip_pickup-1.0.8_v2.diff
( 1) alternative_sip_pickup-1.2.0.diff
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.
Comments:By: Russell Bryant (russell) 2005-11-25 09:43:55.000-0600

+/* Alternative pickup code implemented by */
+/* Martin Pycko (m78pl@yahoo.com) */
+/* ************ */
+/* Enjoy! */

Are you Martin Pycko or did you pay for this code to be written?  Otherwise, we can not consider this code for addition to Asterisk.

By: Russell Bryant (russell) 2005-12-01 00:05:14.000-0600

I'm closing this due to lack of response.  Feel free to re-open to answer the questions that have been posted.  Thanks!

By: kib (kibeki) 2005-12-07 07:16:20.000-0600

We contacted Marcin Pycko, the original write of the patch, and now payed him for updating the code to fit into 1.2.0.
We now want the patch to be integrated to the normal development circuit of asterisk.

By: Serge Vecher (serge-v) 2005-12-07 07:46:39.000-0600

kibeki: all new features will go into the trunk, not 1.2 tree. Right now they are pretty much the same, but that will change with time. Please make sure the patch is against the SVN trunk.

By: kib (kibeki) 2005-12-07 08:29:26.000-0600

vechers: what is the SVN trunk? stupid question, sorry :-(

By: kib (kibeki) 2005-12-08 09:47:07.000-0600

apply the patch like this:
cd asterisk
patch -p0 < ../alternative_sip_pickup-1.2.0.diff

for configuration hints look at: http://www.voip-info.org/wiki/view/Asterisk+bounty+Call+Pickup+with+CID+info.+-+SIP

By: Olle Johansson (oej) 2006-01-04 05:45:35.000-0600

Still no patch to current svn trunk? Will close if no updates are produced soon.


By: kib (kibeki) 2006-01-05 07:30:37.000-0600

I just asked Martin Pycko to do this, because he made the patch.

By: kib (kibeki) 2006-01-26 03:12:26.000-0600

The file alternative_sip_pickup-1.2.0.diff also works for current svn trunk.

By: Olle Johansson (oej) 2006-01-26 03:22:01.000-0600

Who can disclaim this code legally?

By: Olle Johansson (oej) 2006-01-26 03:22:01.000-0600

Who can disclaim this code legally?

By: kib (kibeki) 2006-01-26 06:07:55.000-0600

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.

By: Kevin P. Fleming (kpfleming) 2006-02-14 15:59:47.000-0600

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.

By: kib (kibeki) 2006-02-15 09:03:49.000-0600

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.

By: Olle Johansson (oej) 2006-03-29 19:50:41.000-0600

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!