CallerInfoAsyncQuery can now handle SIP addresses in addition to regular
phone numbers: if the number passed in to startQuery() is actually a "URI
number", we now treat it as a SIP address and look it up directly in the
Data table.
If it's a regular phone number, the behavior is unchanged: we use the
PhoneLookup table as before.
This piece of the fix covers only the contact lookup for incoming calls;
we still need some more cleanup of the CallerInfo class in order to get
the call log working.
Bug: 3004127
Change-Id: I0fcb80f9de5b8ecf99d31ee92e0889ddb07216fd
Merge commit '1738252a596c71851cabf5835acb3584ad6b3191'
* commit '1738252a596c71851cabf5835acb3584ad6b3191':
Don't enter DISCONNECTING state when the call/connection is not alive
and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not
from local exceptions.
http://b/issue?id=3041332
Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
Merge commit '245475925eff61ee76bde58de69253a889e39d0a' into gingerbread-plus-aosp
* commit '245475925eff61ee76bde58de69253a889e39d0a':
Fix the startAudio order for 3-way calls.
Merge commit '3234652242f54e3366e7c74e5a0cf0a7da5871b4' into gingerbread-plus-aosp
* commit '3234652242f54e3366e7c74e5a0cf0a7da5871b4':
Don't enter DISCONNECTING state when the call/connection is not alive
+ check REQUEST_TERMINATED response on INVITE not CANCEL,
+ check if a TransactionTerminatedEvent matches the ongoing transaction,
+ add log to track SipConnection disconnect events.
Change-Id: I28325be62ac44e4a7507d3c4b5b78b066c0ea2ad
and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK
DisconnectCause.
http://b/issue?id=3020185
Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
Merge commit '4a04a3129bd30a996dd302b982aeca8f228f57e8'
* commit '4a04a3129bd30a996dd302b982aeca8f228f57e8':
Fix the unhold issue especially if one is behind NAT.
Merge commit 'bd2294204e3edaede3fe81eb9b11c05c4fafe627' into gingerbread-plus-aosp
* commit 'bd2294204e3edaede3fe81eb9b11c05c4fafe627':
Fix the unhold issue especially if one is behind NAT.
Merge commit 'd6d83279183db749de07bfdac79fe4180fc848d0'
* commit 'd6d83279183db749de07bfdac79fe4180fc848d0':
SIP: longer timeout for making call, shorter for cancelling
Merge commit '194bbcce9ba15634500f542b9ea017b2cf154b45' into gingerbread-plus-aosp
* commit '194bbcce9ba15634500f542b9ea017b2cf154b45':
SIP: longer timeout for making call, shorter for cancelling
Merge commit '84a357bb6a8005e1c5e924e96a8ecf310e77c47c' into gingerbread-plus-aosp
* commit '84a357bb6a8005e1c5e924e96a8ecf310e77c47c':
Refactoring SIP classes to get ready for API review.
+ replace SipAudioCall and its Listener interfaces with real implementations,
+ remove SipAudioCallImpl.java, most of it is has become part of SipAudioCall,
+ add SipSession and its Listener classes to wrap ISipSession and ISipSessionListener,
+ move SipSessionState to SipSession.State,
+ make SipManager keep context and remove the context argument from many methods of its,
+ rename SipManager.getInstance() to newInstance(),
+ rename constant names for action strings and extra keys to follow conventions,
+ set thread names for debugging purpose.
Change-Id: Ie1790dc0e8f49c06c7fc80d33fec0f673a9c3044
+fix the unknown call flash for answering an incoming call and
updating the screen if the background call got dropped.
+change the getFirstActiveBgCall to return the call if the state
is not IDLE. This will help to fix unknown flash if the background
call got dropped.
Change-Id: I9803ccebd919acbd5296e7dfde7dc5f29cc9f180
Merge commit 'ee2ef3220fd27a6332acb2f65951a7fe91e9dfa6' into gingerbread-plus-aosp
* commit 'ee2ef3220fd27a6332acb2f65951a7fe91e9dfa6':
Use PhoneBase in the phone list.
also fix delivering bad news before closing a SipAudioCallImpl object so that
apps can get the current audio-call object state before it's closed:
http://b/issue?id=3009262
Change-Id: I94c19dae8b4f252de869e614ec462b19b4ff2077
