Author Topic: Webrtc phone not hanging up call  (Read 816 times)


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Webrtc phone not hanging up call
« on: August 29, 2016, 17:59:32 »
I'm using Queuemetrics 15.10.6, asterisk 11.21.0 on Cento7 (Elastix 4)
I've managed to get everything up and running with Queuemetrics in terms of the Webrtc phone.
When making a call from the Queuemetrics softphone to another extension (Desk IP phone or X-lite), the call goes through and i'm able to answer with audio both ways. (Perfect so far)
However, If i make a call from Queuemetrics softphone to any other extension and want to Hangup before that person answers I cannot. The call keeps ringing on the other end and I have to refresh my browser page.
When trying this same scenario on the sipml5 webrtc phone, calls can be placed and hangup perfectly. On my debug page on chrome I'm receiving this error:

VM341:1 TypeError: this.o_local_stream.stop is not a function()
tsk_utils_log_error @ VM341:
1tsk_fsm.act @ VM341:1
tsip_dialog.fsm_act @ VM341:3
tsip_dialog.hangup @ VM341:3
tsip_session.__action_handle @ VM341:3
tsip_session.__action_any @ VM341:3
tsip_session.hangup @ VM341:3
SIPml.Session.Call.hangup @ VM341:3
sipHangUp @ VM342:136
sipHangup @ VM343:27
_.Bb @ clirt-1.js:110
_.ld @ clirt-1.js:453
Pl @ clirt-1.js:462
Ol @ clirt-1.js:56
Vl @ clirt-1.js:452
Tl @ clirt-1.js:57
_.ab @ clirt-1.js:59
MH @ clirt-1.js:547
wI @ clirt-1.js:557
(anonymous function) @ clirt-0.js:59
VM341:1 === INVITE Dialog terminated ===

Uncaught Class$S169: Exception caught: (TypeError) : this.o_local_stream.stop is not a function

Any ideas what this issue could relate to?
Thanks in advance