[Freeswitch-users] Compatibility issues with Error Cause: 88 [INCOMPATIBLE_DESTINATION]

[Ján Füri]

Do you have to use Portsip ? FS does webrtc.
Anyway, have you tried media_webrtc=false ?

https://wiki.freeswitch.org/wiki/Variable_media_webrtc

I would try it by :
<action application="bridge" data="{media_webrtc=false}sofia/....

but I'm not sure if it works this way, I've never been trying this before.

Jan





On 23.08.2016 11:00, james wrote:
> Hello:
> I am using portsip webRTC gateway to test the compatibility. I use two 
> phones for webRTC testing. One is normal webRTC client and another one 
> is softphone.
> FreeSWITCH work as a IPPBX. The call flows are:
> the problem is that webRTC calls sip phone without any problem, but 
> sip phone calls to webRTC than failed.
> errors are:
> 2016-08-23 16:29:39.971775 [DEBUG] switch_core_state_machine.c:569 
> (sofia/internal/pu7ai86t at 9t5jv2hi82a5.invalid) State Change 
> CS_REPORTING -> CS_DESTROY
> freeswitch at iZ23lkvsnwpZ>
> 2016-08-23 16:29:39.971775 [DEBUG] switch_core_session.c:1647 Session 
> 171 (sofia/internal/pu7ai86t at 9t5jv2hi82a5.invalid) Locked, Waiting on 
> external entities
> freeswitch at iZ23lkvsnwpZ>
> 2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 
> Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> freeswitch at iZ23lkvsnwpZ>
> 2016-08-23 16:29:39.981772 [NOTICE] switch_ivr_originate.c:2771 Cannot 
> create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION]
> freeswitch at iZ23lkvsnwpZ>
> 2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 
> Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
> freeswitch at iZ23lkvsnwpZ>
> 2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401 Originate Failed. 
>  Cause: INCOMPATIBLE_DESTINATION
> freeswitch at iZ23lkvsnwpZ>
> EXECUTE sofia/internal/1004 at 120.26.130.142 answer()
> -------------------------
> It looks freeswitch send with ICE to webRTC. Does anyone know how i do 
> disable the webRTC negotiation and only send normal sip info to webRTC 
> gateway
>
>
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