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* Keep original phone number in SipConnection.Hung-ying Tyan2010-10-122-6/+17
| | | | | | | | | | In case it's a PSTN number carried by an Internet call, the phone app can still get the original phone number from Connection.getAddress() instead of getting a SIP URI. http://b/issue?id=3085996 Change-Id: Ie6c66100a4b5b2ce3f73baa1b446761cd51d7727
* Merge "Add mock ril control commands and tests" into gingerbreadXia Wang2010-10-112-9/+178
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| * Add mock ril control commands and testsXia Wang2010-10-102-9/+178
| | | | | | | | | | | | | | | | Add mock ril controller commands and test cases: - testStartIncomingCallAndHangup: test start incoming cal and hangup remote - testSetCallTransitionFlag: test call transition flag and call state transition Change-Id: I25ff8ef7931159ef7101b5e8638b9b7438db4f66
* | Don't manually create CallerInfo objects from SipPhoneDavid Brown2010-10-102-47/+11
|/ | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | Currently the SipPhone class manually creates a CallerInfo object, and populates it with very basic info from the SIP address, when making an outgoing call. But this is no longer needed, now that we do caller-id lookup properly for SIP addresses (based on real data from the contacts database -- see bug 3004127 and change https://android-git.corp.google.com/g/70555). And in fact the presence of this initial CallerInfo object actually *disabled* contacts lookup for outgoing calls (bug 3072731). This change removes all that CallerInfo-related stuff from SipPhone. (Thus SipPhone is now consistent with the other phone objects, like GSMPhone and CDMAPhone, in that it doesn't muck with CallerInfo data at all, but instead lets the phone app do it.) Also, update isUriNumber() to handle "%40" in case the passed-in string is URI-escaped. (Nobody depends on that now, but it may be needed in the future, and it's certainly safe to say that "%40" will never be found in a legal PSTN number.) TESTED: - Outgoing SIP call: - In-call UI shows correct contact info - After the call, Call Log shows correct contact info - Incoming SIP call: - In-call UI shows correct contact info - After the call, Call Log shows correct contact info - PSTN calls: - correct contact info everywhere Bug: 3072731 Change-Id: I51434e4e5ad66d2e8ff51fc220001fb74485f0f5
* Fix startDtmf.John Wang2010-10-101-1/+1
| | | | | | | | Call correct startDtmf() function. Bug: 3033030 Change-Id: Ia90311ac5d2e4b070a28533c865c81dc90326557
* Merge "Port mock ril controller and tests to GB. DO NOT MERGE" into gingerbreadXia Wang2010-10-076-0/+486
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| * Port mock ril controller and tests to GB. DO NOT MERGEXia Wang2010-10-076-0/+486
| | | | | | | | Change-Id: Ie58236ecb8648d026356610f429054cb46b8640b
* | Add flag to skip restoring network seletion.John Wang2010-10-071-2/+8
| | | | | | | | | | | | For bug 3045366. Change-Id: I7ad9ff89b0a77df473b638ad917b363c42de5405
* | Changed handling of onQueryCompelete to not use mimeType.Wink Saville2010-10-071-18/+18
|/ | | | | | | | | | Using mimeType causes an IPC request to contacts which can be slow. This can cause an ANR of the Phone app. This change parses the URL and to decide which column to use for the person_id and thus should not cause an ANR. bug: 3060704 Change-Id: I750c72746c7269e162f0338c0a3e00230a600519
* Merge "CallManager/SipPhone: fix reject a ringing call" into gingerbreadHung-ying Tyan2010-10-052-17/+34
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| * CallManager/SipPhone: fix reject a ringing callHung-ying Tyan2010-10-062-17/+34
| | | | | | | | | | | | | | | | | | + CallManager: fix getFirstActiveRingingCall(), getActiveFgCall(), getFirstActiveBgCall() + Set DisconnectCause to be INCOMING_REJECTED when a call is rejected http://b/issue?id=3049671 Change-Id: Ica1d81ca4b71ab0ceb2ab437b82bbb4ccf86fe92
* | Merge changes Ic5551c94,I0207f7f7 into gingerbreadMike Lockwood2010-10-052-0/+7
|\ \ | | | | | | | | | | | | | | | * changes: GPS: Clean up GPS API extension initialization in JNI code. GPS: Add support for AGpsRilInterface.
| * | GPS: Add support for AGpsRilInterface.Miguel Torroja2010-10-052-0/+7
| |/ | | | | | | | | | | | | | | Adding changes to be able to have access to missing data to SUPL (celld, imsi, WAP_PUSH and SMS) Change-Id: I0207f7f7ea6595ed3fd7021cb732feddf52e4cf9 Signed-off-by: Mike Lockwood <lockwood@android.com>
* | Merge "Add supporting MCC=450(Korea)" into gingerbreadRobert Greenwalt2010-10-051-13/+15
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| * Add supporting MCC=450(Korea)Sang-il, Lee2010-10-011-13/+15
| | | | | | | | Change-Id: I009f77b064b8bb7f5bd055ac6e7c55a8f608e9d7
* | SIP: add SERVER_UNREACHABLE error code.Hung-ying Tyan2010-10-052-0/+4
| | | | | | | | | | | | | | | | | | | | Let SipSession return it when UnknownHostException is caught. Add DisconnectCause.SERVER_UNREACHABLE in Connection and have SipPhone report it when receiving SERVER_UNREACHABLE from SipSession. http://b/issue?id=3061691 Change-Id: I944328ba3ee30c0a9386e89b5c4696d4d9bde000
* | SIP telephony cleanup.Hung-ying Tyan2010-10-028-716/+48
| | | | | | | | | | | | | | | | | | + Remove unused classes. + Remove unused imports. + Remove unused code. + add DEBUG flag. Change-Id: Ie1236d909d971093b68b066d3d8c1857ac89f56f
* | Merge "Add CallManager/Phone.setEchoSuppressionEnabled()." into gingerbreadHung-ying Tyan2010-10-015-0/+46
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| * | Add CallManager/Phone.setEchoSuppressionEnabled().Hung-ying Tyan2010-09-305-0/+46
| |/ | | | | | | Change-Id: I7bc6241e6fa815787799a53d6f3a076567edc361
* | Merge "Rub in a little 'ol log-b-gone." into gingerbreadDianne Hackborn2010-09-301-1/+1
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| * | Rub in a little 'ol log-b-gone.Dianne Hackborn2010-09-301-1/+1
| |/ | | | | | | | | | | Mmmmmm... great fresh scent! Change-Id: I050e70b31b5d4a9c6731f15a4b51a3620a33a78d
* | Typo fixes in comments and minor code cleanups.Jake Hamby2010-09-303-7/+7
|/ | | | | | | | | * Fix some typos in Javadoc and log messages. * Remove redundant initializer in BluetoothAdapter.readOutOfBandData() * Use canonical "UTF-8" charset name instead of "UTF8" in BluetoothDevice.convertPinToBytes() Change-Id: I58cd5dc48a7ad0053d204c5f590b4b3d438d8672
* Catch exceptions in SipPhone.canTake().Hung-ying Tyan2010-09-301-8/+26
| | | | | | | | Exceptions may throw during canTake() as the peer may cancel the call and result in a race with this method call. Change-Id: I61903d601d8f9b2dcb4c4fbe1586e2c1a1069109 http://b/issue?id=3033868
* SipPhone: revise hangup() in SipCall and SipConnection.Hung-ying Tyan2010-09-301-11/+21
| | | | | | | | | | | Make them DISCONNECTED immediately. Don't enter DISCONNECTING state and wait until SipSession ends the session. SipSession will get timed out eventually but PhoneApp/user don't need to know this detail and wait. This should fix the bug: http://b/issue?id=3027719 Change-Id: Ida5a1bd09d08b9d591721384b4978127619aab51
* Implement contacts lookup (caller-id) for incoming SIP calls.David Brown2010-09-281-14/+80
| | | | | | | | | | | | | | | | | | 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
* SIP: add DisconnectCause.SERVER_ERRORHung-ying Tyan2010-09-282-1/+4
| | | | | | | | | and fix how SipErrorCode.SERVER_ERROR is determinted from server response, not from local exceptions. http://b/issue?id=3041332 Change-Id: Idce67e29858d5c7573b98b7fa1fac074913d71d6
* Fix the startAudio order for 3-way calls.Chung-yih Wang2010-09-281-2/+2
| | | | Change-Id: Ib387b4b1f641f9bf52dd6007d23aee08f0925811
* Fix build.Hung-ying Tyan2010-09-281-1/+1
| | | | Change-Id: I30f2615bc080db2c672e0391fd8bc735de17fcbf
* Don't enter DISCONNECTING state when the call/connection is not aliveHung-ying Tyan2010-09-281-14/+21
| | | | | | http://b/issue?id=3027719 Change-Id: I1b52418a3695e96b48538fbf14497e34d2cfdda9
* SIP: misc fixesHung-ying Tyan2010-09-281-0/+3
| | | | | | | | + 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
* SipService: handle cross-domain authentication errorHung-ying Tyan2010-09-272-0/+4
| | | | | | | | | and add new CROSS_DOMAIN_AUTHENTICATION error code and OUT_OF_NETWORK DisconnectCause. http://b/issue?id=3020185 Change-Id: Icc0a341599d5a72b7cb2d43675fbddc516544978
* Fix the unhold issue especially if one is behind NAT.Chung-yih Wang2010-09-271-1/+12
| | | | | | +call startAudio() when call is established. Change-Id: Ib6a1e34017fb83007ce275da1991058e8b803833
* Merge "Add debug functions." into gingerbreadJohn Wang2010-09-261-23/+167
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| * Add debug functions.John Wang2010-09-261-23/+167
| | | | | | | | | | | | Enable more debug in CallManager. Change-Id: Iaf66e31dc67f7ca9123fe5d8b351a21b1ec8b987
* | SIP: longer timeout for making call, shorter for cancellingHung-ying Tyan2010-09-251-5/+7
|/ | | | | | http://b/3021865 Change-Id: I354ebcc00f1ac68e4b7b466745c36aeb314f9138
* Refactoring SIP classes to get ready for API review.Hung-ying Tyan2010-09-241-12/+12
| | | | | | | | | | | | | + 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 build.Chung-yih Wang2010-09-211-21/+0
| | | | Change-Id: I6e3f683be22ab834ca1b83a1dbb0b3acd62645d2
* Revert the ANSWERING state.Chung-yih Wang2010-09-214-30/+35
| | | | | | | | | | +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 "Use PhoneBase in the phone list." into gingerbreadJohn Wang2010-09-201-34/+81
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| * Use PhoneBase in the phone list.John Wang2010-09-171-34/+81
| | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | | For bug 3001613. Only use PhoneBase (not PhoneProxy) in CallManager. Both PhoneBase and PhoneProxy implement Phone interface, such as dial(). The real implementation, for example in GSM, is in GSMPhone extending from PhoneBase. So that foregroundCall.getPhone() returns GSMPhone obj. On the other hand, PhoneFactory.getDefaultPhone() returns PhoneProxy obj, which has a class member of GSMPhone. Therefore for phone returned by PhoneFacotry, which is used by PhoneApp, phone.getForegroundCall().getPhone() != phone Change-Id: I8a304098dd86762aaee56fb3c8b76c883e8c9a4f
* | SipPhone: fix missing-call DisconnectCause feedbackHung-ying Tyan2010-09-201-1/+3
| | | | | | | | | | | | | | | | | | 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
* | SIP: convert enum to static final int.Hung-ying Tyan2010-09-201-19/+20
|/ | | | | | Converts SipErrorCode and SipSessionState. Change-Id: Iee3a465649ea89d395b2336bbd673c25113e5f93
* Add debug function.John Wang2010-09-171-5/+56
| | | | Change-Id: If2381b04b08876f78ac93833f848f099de5c6104
* SipPhone: do not append SIP domain to PSTN numberHung-ying Tyan2010-09-171-19/+28
| | | | | | | | in the CallerInfo so that only PSTN number is shown in the call log. http://b/issue?id=2982632 Change-Id: I414f01d16ce64ecb8da7c6943ea7f080bcfd2794
* SipAudioCall: expose startAudio()Hung-ying Tyan2010-09-171-0/+1
| | | | | | so that apps can start audio when time is right. Change-Id: I7ae96689d3a8006b34097533bc2434bc3814b82a
* Add timer to SIP session creation process.Hung-ying Tyan2010-09-171-4/+5
| | | | | | | | | | | | + add timer parameter to ISipSession.make/changeCall(), + add timer paramter to SipAudioCall.make/answer/hold/continueCall()'s, + add timer parameter to SipManager.makeAudioCall(), + modify implementation in SipSessionGroup, SipAudioCallImpl accordingly, + make SipPhone to use it with 8-second timeout. http://b/issue?id=2994748 Change-Id: I661a887e5810087ddc5e2318335e2fa427f80ec6
* Add a new phone state ANSWERING.Chung-yih Wang2010-09-164-4/+23
| | | | | | | | The state ANSWERING is set when we answer an incoming sip call, i.e. sending a 'OK' response to the peer. The state will be set to ACTIVE once the 'ACK' from peer is received. Change-Id: I84ee3cc68222eb34e032896ce23f7431d4ad774a
* Fixing the NPE in SipPhoneChung-yih Wang2010-09-151-1/+1
| | | | | | bug id: http://b/2987816 Change-Id: Iee252eee0a5243b70ff0b6f287279f92235b5b2d
* Merge "SIP: add PEER_NOT_REACHABLE error feedback." into gingerbreadHung-ying Tyan2010-09-142-0/+6
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| * SIP: add PEER_NOT_REACHABLE error feedback.Hung-ying Tyan2010-09-152-0/+6
| | | | | | | | | | | | http://b/issue?id=3002033 Change-Id: Ib64b08919d214acbab89945ac19dc113a68e62ad
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