Wear SIP: audio routing, stock-style dialer, DTMF, incoming-call handling, telecom scaffold

- AudioRouteManager: route to onboard speaker when no BT headset is
  connected, BT SCO when one is; set MODE_IN_COMMUNICATION + focus.
- aaudio: build with AAUDIO_PERFORMANCE_MODE_NONE so Android routing
  (speakerphone) is honoured instead of being bypassed by low-latency.
- DialerScreen: restyled to a stock-dialer layout (large number display,
  3x4 keypad with sub-letters, green call FAB, backspace, recent chips).
- InCallScreen: added DTMF keypad (sendDigit) for voicemail/IVR.
- WearBaresipService: on incoming call, wake screen + vibrate + bring
  activity to foreground so Answer/Decline is visible.
- Fixed dial normalization (bare extension -> full SIP URI with domain)
  and the uap parse in uaEvent so defaultUap is set and calls connect.
- config.static: load g711/opus codecs to resolve 488 Not Acceptable Here.
- Telecom scaffold (experiment): PhoneAccount + ConnectionService handed
  incoming calls via addNewIncomingCall to test stock Wear dialer pickup.
- Submodule libbaresip-android rebuilt with g711/opus/aaudio changes.

Verified: assembleDebug green, installs and registers on Galaxy Watch,
incoming call event fires and is handled without crashing.

wear/src/main/assets/config.static

@@ -0,0 +1,42 @@
+poll_method epoll
+call_local_timeout 60
+call_max_calls 4
+call_hold_other_calls yes
+filter_registrar udp,tcp,tls,ws,wss
+audio_player aaudio,nil
+audio_source aaudio,nil
+audio_alert aaudio,nil
+audio_level no
+ausrc_format s16
+auplay_format s16
+auenc_format s16
+audec_format s16
+audio_buffer 20-160
+audio_silence -35.0
+audio_telev_pt 101
+audio_jitter_buffer_type adaptive
+audio_jitter_buffer_ms 100-200
+audio_jitter_buffer_size 50
+rtp_stats no
+rtp_timeout 60
+rtp_rxmode thread
+module aaudio.so
+module g711.so
+module opus.so
+module stun.so
+module turn.so
+module ice.so
+module srtp.so
+module dtls_srtp.so
+module gzrtp.so
+module uuid.so
+module_app account.so
+module_app debug_cmd.so
+module_app mwi.so
+opus_samplerate 16000
+opus_stereo no
+opus_sprop_stereo no
+opus_cbr no
+opus_inbandfec yes
+opus_application voip
+dtls_srtp_use_ec prime256v1