Hi. Trying to get SIP card working. I installed a FreePBX, enabled ws/wss transport, accepted certificate and added extensions. Then I added a card and tried to ring from microsip (6006 not available), but asterisk -rvvvvv shows:
== WebSocket connection from '10.10.100.102:50528' for protocol 'sip' accepted using version '13'
-- Added contact 'sip:3t96thn9@10.10.100.102:50528;transport=ws' to AOR '6006' with expiration of 600 seconds
== WebSocket connection from '10.10.100.102:50509' closed
-- Contact 6006/sip:3t96thn9@10.10.100.102:50528;transport=ws is now Unreachable. RTT: 0.000 msec
........
-- Executing [s@macro-dial-one:61] NoOp("PJSIP/6001-00000002", "Returned from dial-one with nothing to call and DIALSTATUS: CHANUNAVAIL") in new stack
And then nothing server keep sending options and there is no messages from sip card.
Asterisk version: 17.9.4
HA version: 2023.7.2
SIP card version: 2.4.0
Ok. This is because of docker-packed asterisk. I sends its own internal rtp address. And card client doesn't reply with OK.
Tried network_mode=host and this helped.
Hi. Trying to get SIP card working. I installed a FreePBX, enabled ws/wss transport, accepted certificate and added extensions. Then I added a card and tried to ring from microsip (6006 not available), but asterisk -rvvvvv shows:
Chrome F12 dev tool shows ws connection with:
And then nothing server keep sending options and there is no messages from sip card. Asterisk version: 17.9.4 HA version: 2023.7.2 SIP card version: 2.4.0
Am I doing something wrong?