[asterisk-dev] SIP Missed Calls when calling SIP phones parallel

[Jared Smith]

On 6/21/07, Olle E Johansson <olle at voop.com> wrote:
> Yes, this is my code. But please check out the branch and try it out
> instead of sending
> out random patches. I've worked with Frank Sautter who has been
> helping me test
> this and I have implemented other changes since the code you send out.

For everyone else's benefit, I'll post the URL where people can either
look at the code:

http://svn.digium.com/view/asterisk/team/oej/cancel_answer_elsewhere/

or check it out from Subversion:

http://svn.digium.com/view/asterisk/team/oej/cancel_answer_elsewhere/

Please check it out and give feedback to Olle.

-Jared