Para master asterisk, saya ada kendala pengaturan asterisk webrtc dan webclient voip menggunakan JsSIP
untuk pengaturan webRTC saya sudah melihat di artikel ini Configuring Asterisk for WebRTC Clients - Asterisk Documentation
utk pengaturan jssip nya seperti yg ada di link berikut
tetapi setelah saya run tidak ada panggilan. berikut log di asterisk nya
== WebSocket connection from ‘192.168.43.17:52856’ for protocol ‘sip’ accepted using version ‘13’
– Added contact ‘sip:q717l0a5@192.168.43.17:52856;transport=ws;x-ast-orig-host=92j16gen600j.invalid:0’ to AOR ‘1000’ with expiration of 600 seconds
== Endpoint 1000 is now Reachable
[Oct 2 13:57:11] ERROR[2856]: iostream.c:647 ast_iostream_start_tls: Problem setting up ssl connection: error:00000001:lib(0)::reason(1), Internal SSL error
[Oct 2 13:57:11] ERROR[2856]: tcptls.c:179 handle_tcptls_connection: Unable to set up ssl connection with peer ‘192.168.43.17:52840’
[Oct 2 13:57:11] ERROR[2856]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0)::reason(1), Internal SSL error
[Oct 2 13:57:11] NOTICE[2828]: res_pjsip_session.c:4022 new_invite: 1000: Call (WSS:192.168.43.17:52856) to extension ‘6001’ rejected because extension not found in context ‘default’.
kira2 salah dimananya ya
dialcall sudah bisa dari web browser ke softphone. tp saat test audio/speaker tidak ada suara di mic atau di audio/speaker
berikut log webrtc
== WebSocket connection from ‘192.168.43.17:44760’ for protocol ‘sip’ accepted using version ‘13’
– Added contact ‘sip:s7v0q7n2@192.168.43.17:44760;transport=ws;x-ast-orig-host=9dhc1ptap738.invalid:0’ to AOR ‘1000’ with expiration of 600 seconds
== Endpoint 1000 is now Reachable
– Executing [6001@test:1] Answer(“PJSIP/1000-00000002”, “”) in new stack
> 0x7fdb38056020 – Strict RTP learning after remote address set to: 192.168.43.17:37199
> 0x7fdb38056020 – Strict RTP learning after ICE completion
> 0x7fdb38056020 – Strict RTP learning after remote address set to: 192.168.43.17:37199
> 0x7fdb38056020 – Strict RTP switching to RTP target address 192.168.43.17:37199 as source
– Executing [6001@test:2] Dial(“PJSIP/1000-00000002”, “PJSIP/6001,60”) in new stack
– Called PJSIP/6001
– PJSIP/1000-00000002 requested media update control 26, passing it to PJSIP/6001-00000003
– PJSIP/1000-00000002 requested media update control 26, passing it to PJSIP/6001-00000003
– PJSIP/6001-00000003 is ringing
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [6001@test:3] Playback(“PJSIP/1000-00000002”, “vm-nobodyavail”) in new stack
– <PJSIP/1000-00000002> Playing ‘vm-nobodyavail.g722’ (language ‘en’)
> 0x7fdb38056020 – Strict RTP learning complete - Locking on source address 192.168.43.17:37199
– Auto fallthrough, channel ‘PJSIP/1000-00000002’ status is ‘BUSY’
apa ada yang kurang pengaturan di asterisk nya atau di sisi jssip nya