webrtc + fusionpbx, webphones can’t call sip phones or trunks

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CrazyTux

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Jun 15, 2021
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web phones call each other, but it is not possible to call from them or to them from vulture devices, the call ends with Originate Failed. Cause: INCOMPATIBLE_DESTINATION
in the logs it can be seen that the rtp is transmitted as is, without coordination and transcoding
As a result, srtp goes to clients where there is no support for it or codecs that are not supported by the client are used. This does not happen with ordinary vulture clients. If anyone can tell you where to get this monster to take a call along with the media and send it in transcoded form further, I will be very grateful.
I do not have such probes in the asterisk, but I cannot use it in this case :(
 
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