[Freeswitch-users] drops calls in the first 60 seconds

[Brian West]

Get sip traces of it not working and revert your sofia profiles back to what is in git and I bet it will just work.  

So you can take your pick.. make your device do its job and not force the registrar to overcome your devices stupidness and you can have call density out the wazzo or you can pick to have FS overcome it for them chewing up more cpu and processing power trying to overcome your stupid device and then only have a handful of calls on a machine.  Its really what it boils down to.

/b

On Dec 9, 2010, at 12:57 PM, Aloysius Lloyd wrote:

> Brian,
> 
> In the current scenario Asterisk and Aastra phones working smoothly without any additional settings.
> All I need to enter username , password and sip server address.
> 
> But FreeSWITCH and Aastra phone - I need to enter username , password , sip server address , rport enable. Without rport enabled, phone never register to FreeSSWITCH.
> 
> Why FreeSWITCH and Nat behaving like this? What is difference between FreeSWITCH NAT and Asterisk NAT.
> 
> I do not want to go back Asterisk. I want to educate myself. Also want to switch more customers to FreeSWITCH.
> 
> Any help is appreciated.
> 
> Thanks
> Lloyd

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