I strongly reccomend using FreeSWITCH 1.6 for this. 1.4 is no longer receiving any updates except for critical secuity fixes and significant work has been done on the area you are having issues with in 1.6.<span></span><br><br>On Thursday, July 28, 2016, Sangram Rath <<a href="mailto:s.rath@voverc.com">s.rath@voverc.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Hi Group,<br><br>We have an issue here where we are not able to make inbound/outbound calls using a websocket application that uses sipjs.<br>We think that wan.auto is not excluding IPv6. This problem is related to ICE. There was a bug on this that was fixed in 1.4.19.<br><br>Additional info:<br></div>- The platform is Kazoo<br><div>- FS version: 1.4.23 (also tried upgrading to 1.4.26)<br>- IPv6 is disabled in the OS<br><br>We have the below in logs:<br>[WARNING] switch_core_media.c:2790 NO candidate ACL defined, Defaulting to wan.auto<br>[DEBUG] switch_core_media.c:5178 AUDIO RTP [sofia/sipinterface_1/<a href="javascript:_e(%7B%7D,'cvml','DPRD025488@somedomain.com');" rel="nofollow" target="_blank">DPRD025488@somedomain.com</a>] x.x.x.x port 24092 -> 2001::5ef5:79fd:207f:3d43:3f57:fe92 port 61975 codec: 8 ms: 20<br>[DEBUG] switch_rtp.c:3588 Not using a timer<br>[INFO] switch_core_media.c:5352 Activating Audio ICE<br>[NOTICE] switch_rtp.c:4029 Activating RTP audio ICE: bveNJceaOHdVG0Tq:tThuJX7NS3O8dtZ1 2001::5ef5:79fd:207f:3d43:3f57:fe92:61975<br>[INFO] switch_core_media.c:5395 Activating RTCP PORT 61975<br>[DEBUG] switch_rtp.c:3929 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 61975<br>[DEBUG] switch_rtp.c:2349 Setting RTCP remote addr to :61975<br>[INFO] switch_core_media.c:5403 Skipping RTCP ICE (Same as RTP)<br>[INFO] switch_rtp.c:3104 Activate RTP/RTCP audio DTLS client<br><br>It would be great to have ideas on what could be the reason.<br><br>Thanks,<br>Sangram<br clear="all"><br></div></div>
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