> Hi all, > > I'm running the following: > > > sipXconfig (4.1.0-016351 2009-09-02T16:01:46 ecs-centos5.x64) > > > I'm having an issue where I can't call my registered UA's. > The INVITE just never hits the phone. Outbound calls work fine. > > > Here is an example of my registered Polycom UA: > > <sip:[email protected]:50390;x-sipX-privcontact=192.168.1.47> > > So in the above case, my external IP is 66.100.226.81 and the > internal IP of the UA is 192.168.1.47. > > Now when I call the phone the sipXecs server on a static > public IP sends the INVITE to the internal IP. So the INVITE > never reaches the UA. > > > Here are some headers of the INVITE: > > U 2009/10/03 16:15:13.843453 201.11.178.41:5060 -> > 192.168.1.47:5060 INVITE sip:[email protected] > <mailto:sip%[email protected]> SIP/2.0. > Record-Route: > <sip:201.11.178.41:5060;lr;sipXecs-CallDest=INT;sipXecs-rs=%2A auth%7E.%2Afrom%7ENTA3REU4QkItMTY1OERBQTA%60%> 21fc4fcff96f8918df58248a2b61ecf8fa>. > > > Why would this be happening? > > Perhaps I'm missing a config setting?
That is an interesting problem you are having. Based on the information you have supplied, it would seem like sipXecs should have routed the INVITE to the public IP address of the phone. I looked at the code path and the only plausible explanation for what you are observing is that the NAT Traversal feature is not running. Please verify the following: In sipXconfig: System->Internet Calling->NAT Traversal->Enable NAT Traversal checkbox: verify that it is 'checked' In sipXconfig: System->Servers->[click on your server]: verify that Media Relay is running If the two things above are ok then try this: In sipXconfig: System->Servers->[click on your server]: click the checkbox to the left of SIP Proxy and hit 'Restart' then wait for 2 minutes and go check if you have any alarms in Diagnostics->Alarms->History that would say anything about NAT Traversal failing to initialize. To get to the bottom of this, I will need a snapshot from you. To that end, go to System->Logging Levels->Show advanced settings and set the SIP Proxy logging level to DEBUG. Once this is done, restart the SIP Proxy following the steps already described above. Once restarted, give it two minutes and then attempt a single failing call to Polycom. Finally, take a snapshot of the system under sipXconfig->Diagnostics->Snapshot and e-mail it to me. What you are trying to achieve is definitely supported and is being used on a daily basis by many of us in our office. We'll get to the bottom if it... bob _______________________________________________ sipx-users mailing list [email protected] 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/
