[asterisk-dev] SIP transfer committed

[Johansson Olle E]

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