Open prathibhatvm opened 2 years ago
30 seconds sounds suspiciously like a default timeout of sorts. Doesn’t something appear in the Asterisk CLI at this point?
== WebSocket connection from '10.176.16.37:55910' for protocol 'sip' accepted using version '13' == WebSocket connection from '10.176.16.37:55911' for protocol 'sip' accepted using version '13' -- Registered SIP 'vinod' at 10.176.16.37:55910
Saved useragent "Raspberry Phone (JsSIP - 0.20.0)" for peer vinod -- Registered SIP 'prathibha' at 10.176.16.37:55911 Saved useragent "Raspberry Phone (JsSIP - 0.20.0)" for peer prathibha [Feb 22 11:36:13] NOTICE[2457]: chan_sip.c:24640 handle_response_peerpoke: Peer 'vinod' is now Reachable. (55ms / 2000ms) [Feb 22 11:36:13] NOTICE[2458]: chan_sip.c:24640 handle_response_peerpoke: Peer 'prathibha' is now Reachable. (16ms / 2000ms) [Feb 22 11:36:14] NOTICE[2457]: chan_sip.c:28463 handle_request_subscribe: Received SIP subscribe for peer without mailbox: vinod [Feb 22 11:36:14] NOTICE[2458]: chan_sip.c:28463 handle_request_subscribe: Received SIP subscribe for peer without mailbox: prathibha == DTLS ECDH initialized (automatic), faster PFS enabled == DTLS ECDH initialized (automatic), faster PFS enabled == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 0x7f8e7c01c420 -- Strict RTP learning after remote address set to: 14.139.183.221:62271 -- Executing [10100@from-extensions:1] Gosub("SIP/vinod-00000000", "dial-extension,s,1,(prathibha)") in new stack -- Executing [s@dial-extension:1] NoOp("SIP/vinod-00000000", "Calling: prathibha") in new stack -- Executing [s@dial-extension:2] MixMonitor("SIP/vinod-00000000", "voice-10105-prathibha-Tue Feb 22 06:06:58 2022.wav") in new stack -- Executing [s@dial-extension:3] Dial("SIP/vinod-00000000", "SIP/prathibha,30") in new stack == Begin MixMonitor Recording SIP/vinod-00000000 == DTLS ECDH initialized (automatic), faster PFS enabled == DTLS ECDH initialized (automatic), faster PFS enabled 0x7f8e7c03f1b0 -- Strict RTP learning after ICE completion == Using SIP VIDEO CoS mark 6 [Feb 22 11:37:21] NOTICE[2367]: chan_sip.c:28692 handle_request_register: Registration from 'sip:400@10.176.16.170' failed for '10.176.16.57:5060' - Wrong password == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/prathibha -- SIP/prathibha-00000001 is ringing [Feb 22 11:37:39] WARNING[2549][C-00000000]: translate.c:407 framein: no samples for opustolin
while starting the video call, the user is getting unregistered.
quit_handler: write() failed: Bad file descriptor
Starting Video Call... After 32 secs, on user1's end it gets disconnected. On the other end, again after 30 secs, the call appears. On clicking, 'Answer with Video Call', the calls starts with 1 user.