[asterisk-dev] SipAddHeader and SIP REFER

[Olle E. Johansson]

15 mar 2011 kl. 16.35 skrev Russell Bryant:

> On Tue, 2011-03-15 at 09:01 +0100, Olle E. Johansson wrote:
>> 1) Be careful with locking issues. I added something that checked channel variables during a call (MAX_FORWARDS) that we simply could not solve.
>> If you want to use this for a REFER during a call, you're in the same neighbourhood. That's no good.
> 
> You're right that this is something to watch out for.  We can make sure
> it's safe through peer code review though.

Since we found no solution to this, but reverted backwards and avoided the problem - do you have any advice on how
to solve this so we don't put this developer on a bad track from start. That's not very nice :-)

> 
>> 2) Maybe we should consider a separate list of headers for REFER. THe proper way would be to store these in a channel storage unit, not in channel variables. The current sip-header implementation was done before we had data storage for channels.
> 
> Agreed that a channel datastore would be a better storage mechanism than
> channel variables.  I like this option of adding it as something that
> you have to explicitly say you want on REFER.

If that works, we can later add a similar function for BYE, something a lot of people have asked for.

/O