Compatibility issues with Error Cause: 88 [INCOMPATIBLE_DESTINATION]
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> EXECUTE sofia/internal/1004@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
Freeswitch already supports webrtc, there is no need for a gateway.
On Aug 23, 2016, at 5:00 AM, james <freepbx@qq.com> 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: <290CF6FD@45AAC238.9810BC57> 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> EXECUTE sofia/internal/1004@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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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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> EXECUTE sofia/internal/1004@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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> EXECUTE sofia/internal/1004@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Guillermo Ruiz Camauer
Yep, codecs: <!-- Force that PCMU codec on Leg B --> <execute application="export" data="nolocal:absolute_codec_string=PCMU"></execute> On Tue, Aug 23, 2016 at 1:24 PM, Guillermo Ruiz Camauer <grcamauer@gmail.com
wrote:
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> EXECUTE sofia/internal/1004@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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.orghttp://www.freeswitchsolutions.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon....
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Guillermo Ruiz Camauer
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Sincerely, Don Hawkins CEO Hawkins Enterprise Group LLC http://hawkinsegroup.com Zello PTT <http://zello.com>: push2don P: 469-214-5044
Thank you, I will try : action application="bridge" data="{media_webrtc=false}sofia/.... ------------------ Original ------------------ From: "james";<freepbx@qq.com>; Date: Tue, Aug 23, 2016 05:00 PM To: "freeswitch-users"<freeswitch-users@lists.freeswitch.org>; Subject: Compatibility issues with Error Cause: 88 [INCOMPATIBLE_DESTINATION] 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> EXECUTE sofia/internal/1004@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
participants (5)
-
Don Hawkins
-
Guillermo Ruiz Camauer
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james
-
Ján Füri
-
Michael Jerris