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Re: [LUG] SIP calls via Asterisk through NAT

 

On 10/09/13 10:39, Rob Beard wrote:
The phones are Grandstream ones, can't remember the exact model number off the top of my head.

I wasn't sure if the port forwarding should point to the Asterisk server or the phones, I was guessing the Asterisk server.

yes the * server.
Have a look at the canreinvite option in sip.conf for the sipgate entry, mine are no

One thing to be aware of if you lose access to DNS SIP locks up * (I.E. internet dies). There is a bug/feature in the SIP code that waits for DNS resolution and has too long a time out. I run two * servers on the same box to get around the problem, one is internal and does phones and POTS and the other does SIP to outside world, they are trunked together. Might have been fixed in more recent versions of * but still exists in 1.6.2.7 I am using.

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