Hi, we have installed 4.0.2-016420 on Centos 5. We have applied latest patch for sipXbridge too (4.0.4).
We are experience a problem with audio on our calls. When we do internal calls from extension to extension phones are ringing, but we don't have audio when we pickup the phones. In this case we don't have set STUN on the phones. Especially this happens using hardware over IP phones. Using softphones seems to be ok. When we set STUN on hardware phone and the others, then the voice is comming back anad everything is ok. This problem appear if we call to a real number and on the line is put hardware voip phone. How to deal with this problem? I don't see any problems in the logs. I know that this is related with NAT Traversal. Server is not behind NAT and it has properly configured DNS records, and it uses static IP. My NAT settings on sipxecs are this way: System -> Internet Calling -> NAT Traversal -> Enable NAT Traversal (checked), Server behind NAT (not checked). System -> Servers -> choose the server -> NAT -> Specify IP Address. ( as i understand, this option is related to way if Server is behind NAT or not, or if server has dynamic address. I suppose that this setting does not have any relation with remote workers behind NAT). Is there a way to set in sipxecs remote workers which stun server to use, or something like this or there is another solution? So how to go with this problem? I found this thread: http://www.mail-archive.com/sipx-users@list.sipfoundry.org/msg05291.html My problem is the same or very similar. What i expect is to not be necessary to set on the phones (hardware, softphone, etc.) stun. I suppose that this functionality must be covered from sipXbridge, but maybe i'm wrong. If you need log parts, please tell me and i will post logs here. Thanks in advanced! _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
