A few fixes to SIP with regards to connected line updates during transfers.
* Set the invitestate to INV_CALLING when we send a connected line reinvite.
This prevents us from potentially rapid-firing reinvites to a single peer.
* Use the astdb to store a peer's allowed methods. This prevents us from sending
an UPDATE during the interval between startup and the peer's first registration
if the peer does not support the UPDATE method.
* Handle Polycom's method of indicating allowed methods in REGISTER. Instead of
using an Allow header, they place the allowed methods in a methods= parameter
in the Contact header.
ABE-1873
........
Modified:
branches/1.6.2/ (props changed)
Propchange: branches/1.6.2/
------------------------------------------------------------------------------
Binary property 'trunk-blocked' - no diff available.
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum