-
- Downloads
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Please register or sign in to comment