The UDP switch made the watch register over UDP while Asterisk had
cached a TCP Contact (transport=tcp). Asterisk sent the INVITE over
TCP but got no SIP response (not even 100 Trying) -> 6.4s
retransmission timeout -> immediate voicemail.
Revert to TCP so the watch's registration matches Asterisk's cached
contact. With nat=yes now set, the TCP Contact is made public
(78.141.29.181:<port>), so the INVITE arrives over the persistent TCP
socket and baresip's response reaches Asterisk, letting the call ring
the full configured window instead of timing out.
Switch outbound transport from tcp to udp. With Asterisk peer
nat=yes, Asterisk now sends the INVITE to the public source address
(78.141.29.181:<port>) instead of the watch's private Contact, so
incoming calls traverse the NAT and reach the watch.
Verified: external incoming calls now arrive (uaEvent: incoming call
from external caller logged on the watch).
- Account line now uses 01273961147 (same credential).
- sipnat=outbound + natpinhole=yes + regint=60 keep the NAT
mapping/keep-alive fresh so Asterisk's INVITE (sent to the public
Contact) reaches the watch behind NAT.
Note: sipnat=outbound is ineffective here because the outbound proxy
is the server itself (no edge STUN to learn the watch's public NAT
IP); the actual NAT fix is Asterisk-side (nat=force_rport,comedia on
the peer), which makes Asterisk target the public source address.
- 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.
