1 jun 2006 kl. 16.51 skrev Mike Fedyk:
Olle E Johansson wrote:
Friends,
I finally committed the last piece of the new SIP transfer support
code. This greatly enhances the support of SIP
transfers - or at least is meaning to. The code has been tested on
1.2 for almost a year in production, but the
trunk version is a port from this. A port in many cases means that
one introduces new bugs.
How were transfers done on SIP before this change?
Well, we had a SIP transfer manager in Jönköping that took care of
all of that.
Just joking. It's a long story, but things that did not work properly
- Transfers between two servers
- Transfers to bad extensions (the transfer target got the errors and
we told the transferer that it was ok)
- Transfers of calls in early state (ringing)
And a lot of minor stuff. Basically, we handle transfers much more
properly and a lot of people will be
surprised by Asterisk suddenly responding with a failure to the phone
and keeping the call up.
Also, you can now disable SIP transfers totally or per peer/user in
sip.conf. There was no way you
could do that before. SIP transfers was always accepted.
/O_______________________________________________
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