jamie prince
300bdcc810
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.
3 lines
194 B
Plaintext
3 lines
194 B
Plaintext
<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
|
|
|