I believe he is just trying to test the gateway.
Most of the time that I get the INCOMPATIBLE DESTINATION error it is because there are no common CODECS.  Are you sure you have the right codecs enabled in Freeswitch?

Regards,

Guillermo

On Tue, Aug 23, 2016 at 11:45 AM, Ján Füri <furi@vmtele.com> wrote:
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@9t5jv2hi82a5.invalid) State Change CS_REPORTING -> CS_DESTROY
freeswitch@iZ23lkvsnwpZ> 
2016-08-23 16:29:39.971775 [DEBUG] switch_core_session.c:1647 Session 171 (sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid) Locked, Waiting on external entities
freeswitch@iZ23lkvsnwpZ> 
2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
freeswitch@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@iZ23lkvsnwpZ> 
2016-08-23 16:29:39.981772 [DEBUG] switch_ivr_originate.c:3759 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION]
freeswitch@iZ23lkvsnwpZ> 
2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401 Originate Failed.  Cause: INCOMPATIBLE_DESTINATION
freeswitch@iZ23lkvsnwpZ> 
-------------------------
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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_________________________________________________________________________
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consulting@freeswitch.org
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Official FreeSWITCH Sites
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--
Guillermo Ruiz Camauer