Wear SIP: revert to TCP registration (fixes INVITE retransmit timeout)

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.
This commit is contained in:
2026-08-20 23:19:39 +01:00
parent 34661f4ca6
commit 300bdcc810

View File

@ -1,2 +1,2 @@
<sip:01273961147@mail.txt3.net>;auth_user="01273961147";auth_pass="cisco55555";outbound="sip:mail.txt3.net:5060;transport=udp";regint=60;regq=0.5;pubint=0;check_origin=no;mwi=no;natpinhole=yes <sip:01273961147@mail.txt3.net>;auth_user="01273961147";auth_pass="cisco55555";outbound="sip:mail.txt3.net:5060;transport=tcp";regint=60;regq=0.5;pubint=0;check_origin=no;mwi=no;natpinhole=yes