[asterisk-dev] SIP Missed Calls when calling SIP phones parallel
linux at nowin.de
Thu Jun 21 16:14:46 CDT 2007
In the mentioned thread in this list I told about the check on
p->owner. Look here:
You changed this in your branch. The branch works, my colleague made a
short test regarding the missed calls.
But as also told trunk is no choice for my office. We need a stable
Bristuffed-Asterisk and so I backported your changes to our rock-solid
Asterisk 1.2. I know Asterisk-developers don't like that.
In the feature I plan to upgrade to Asterisk 1.6. But it has to be
stable and there have to be a Bristuffed version, cause I need the
Thursday, June 21, 2007, 10:07:16 PM, you 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.
> Getting feedback is important. If you change the code to work and not
> contribute back
> to me, how should I know that it's wrong in your network?
> Please test the branch and provide me with patches, feedback. Thanks.
> --Bandwidth and Colocation provided by Easynews.com --
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
More information about the asterisk-dev