Hi,

first my FS is on public ip.

We have migrated  from 1.2.x to Version
1.4.8+git~20140910T181155Z~9f72b11892~64bit (git 9f72b11 2014-09-10
18:11:55Z 64bit as we needed different server.

I have noticed that a lot of costumers are complaining that a lot of
phone call has been cut in about 1:30 min of the call. From what I can
see from sip trace is that FS begins to send media to private ip.

link to trace: http://pastebin.com/9M6X6SFp

So, rtp is send first to right ip and media is ok but in about 1:30 min
of the call the FS sends update and begin to send rtp to private ip.


thx for help!

miha

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