<div dir="ltr">I believe he is just trying to test the gateway.<div>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?</div><div><br></div><div>Regards,</div><div><br></div><div>Guillermo</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Aug 23, 2016 at 11:45 AM, Ján Füri <span dir="ltr"><<a href="mailto:furi@vmtele.com" target="_blank">furi@vmtele.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
Do you have to use Portsip ? FS does webrtc.<br>
Anyway, have you tried media_webrtc=false ? <br>
<br>
<a href="https://wiki.freeswitch.org/wiki/Variable_media_webrtc" target="_blank">https://wiki.freeswitch.org/<wbr>wiki/Variable_media_webrtc</a><br>
<br>
I would try it by :<br>
<action application="bridge" data="{media_webrtc=false}<wbr>sofia/....<br>
<br>
but I'm not sure if it works this way, I've never been trying this
before.<br>
<br>
Jan<div><div class="h5"><br>
<br>
<br>
<br>
<br>
<br>
<div>On 23.08.2016 11:00, james wrote:<br>
</div>
</div></div><blockquote type="cite"><div><div class="h5">
<div>Hello:</div>
<div>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.</div>
<div>FreeSWITCH work as a IPPBX. <span style="line-height:1.5">The
call flows are:</span></div>
<div><span style="color:rgb(204,0,0);font-family:Arial,Helvetica;font-size:13px;line-height:18px"></span></div>
<div><img src="cid:part1.F1E22F0B.70F6EC1A@vmtele.com"> </div>
<div>the problem is that webRTC calls sip phone without any
problem, but sip phone calls to webRTC than failed. </div>
<div>errors are:</div>
<div>
<div>2016-08-23 16:29:39.971775 [DEBUG]
switch_core_state_machine.c:<wbr>569
(<a href="mailto:sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid" target="_blank">sofia/internal/pu7ai86t@<wbr>9t5jv2hi82a5.invalid</a>) State Change
CS_REPORTING -> CS_DESTROY</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.971775 [DEBUG]
switch_core_session.c:1647 Session 171
(<a href="mailto:sofia/internal/pu7ai86t@9t5jv2hi82a5.invalid" target="_blank">sofia/internal/pu7ai86t@<wbr>9t5jv2hi82a5.invalid</a>) Locked, Waiting
on external entities</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [DEBUG]
switch_ivr_originate.c:3759 Originate Resulted in Error Cause:
88 [INCOMPATIBLE_DESTINATION]</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [NOTICE]
switch_ivr_originate.c:2771 Cannot create outgoing channel of
type [user] cause: [INCOMPATIBLE_DESTINATION]</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [DEBUG]
switch_ivr_originate.c:3759 Originate Resulted in Error Cause:
88 [INCOMPATIBLE_DESTINATION]</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>2016-08-23 16:29:39.981772 [INFO] mod_dptools.c:3401
Originate Failed. Cause: INCOMPATIBLE_DESTINATION</div>
<div>freeswitch@iZ23lkvsnwpZ> </div>
<div>EXECUTE <a href="mailto:sofia/internal/1004@120.26.130.142" target="_blank">sofia/internal/1004@120.26.<wbr>130.142</a> answer()</div>
</div>
<div>-------------------------</div>
<div>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<span style="color:rgb(118,118,118);font-family:Arial,Helvetica;font-size:13px;line-height:20px"></span></div>
<br>
<fieldset></fieldset>
<br>
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