Unable to place calls with WebRTC

I am running the latest version of freepbx and just discovered the UCP and WebRTC. I’ve enabled the WebRTC using my default certificate that was generated at FreePBX Distro install for testing, and I can attempt to dial, but when I click call (and then allow Chrome to use my mic) it immediately hangs up.

My Asterisk Logfiles Show quite a bit of errors and warnings similar to the following:

[2014-11-25 16:06:56] WARNING[7970][C-00000010] chan_sip.c: Rejecting secure audio stream without encryption details: audio 34692 RTP/SAVPF 111 103 104 0 8 106 105 13 126
[2014-11-25 16:11:55] VERBOSE[7970] res_http_websocket.c: == WebSocket connection from '50.243.31.253:47907' closed
[2014-11-25 16:12:56] VERBOSE[4475] chan_sip.c: -- Registered SIP '7117' at 10.0.0.149:43686
[2014-11-25 16:12:56] VERBOSE[4475] chan_sip.c: -- Registered SIP '7117' at 10.0.0.149:6
[2014-11-25 16:17:56] VERBOSE[9204] res_http_websocket.c: == WebSocket connection from '10.0.1.82:61403' for protocol 'sip' accepted using version '13'
[2014-11-25 16:17:56] NOTICE[9204] chan_sip.c: Registration from '<sip:[email protected]>' failed for '10.0.1.82:61403' - Wrong password
[2014-11-25 16:18:14] VERBOSE[9204][C-00000011] netsock2.c: == Using SIP RTP TOS bits 184
[2014-11-25 16:18:14] VERBOSE[9204][C-00000011] netsock2.c: == Using SIP RTP CoS mark 5
[2014-11-25 16:18:14] NOTICE[9204][C-00000011] chan_sip.c: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 35323 RTP/SAVPF 111 103 104 0 8 106 105 13 126
[2014-11-25 16:18:14] WARNING[9204][C-00000011] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35323 RTP/SAVPF 111 103 104 0 8 106 105 13 126

Any thoughts? I’m assuming the “997117” is the auto-generated extension for WebRTC (my extension is 7117)?

The SIP registration for 7117 shown is my desk phone.

I’m new to this, but thought it would be pretty sweet to get WebRTC running.

Thanks!

I’m experiencing the same problem, using Asterisk 11.14.2 and FPBX 12.0.19

   -- Executing [[email protected]:22] Dial("SIP/99411-000004da", "SIP/SIPBOUND/mycellphone,300,Tt") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Called SIP/SIPBOUND/mycellphone
       > 0x7f644150f050 -- Probation passed - setting RTP source address to ipaddress:17076
       > 0x7f644150f050 -- Probation passed - setting RTP source address to ipaddress:17076
    -- SIP/SIPBOUND-000004db is ringing
    -- SIP/SIPBOUND-000004db is making progress passing it to SIP/99411-000004da
       > 0x7f644150f050 -- Probation passed - setting RTP source address to ipaddress:17076
    -- SIP/SIPBOUND-000004db answered SIP/99411-000004da
    -- Executing [[email protected]:1] Macro("SIP/99411-000004da", "hangupcall,") in new stack
    -- Executing [[email protected]:1] GotoIf("SIP/99411-000004da", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [[email protected]:3] ExecIf("SIP/99411-000004da", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [[email protected]:4] Hangup("SIP/99411-000004da", "") in new stack
  == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/99411-000004da' in macro 'hangupcall'
  == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/99411-000004da'
  == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/99411-000004da' in macro 'dialout-trunk'
  == Spawn extension (restrictedroute-1, mycellphone, 6) exited non-zero on 'SIP/99411-000004da'

Any thoughts?