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On Tue, 10 Sep 2013, stinga wrote:
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
It can never reinvite if there is NAT in the way. There are some ways to trick it, but they're fraught with NAT issues all the way.
SIP was developed at the same time as NAT was being developed (late 90's) It's such a shame they never thought about it better. There are other VoIP protocols that interoperate much better with NAT - e.g. IAX, (& Skype) but the number of desk phones that support IAX can be counted with the thumbs on your hands.
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.
Run a local caching DNS server. Or use numbers. Gordon -- The Mailing List for the Devon & Cornwall LUG http://mailman.dclug.org.uk/listinfo/list FAQ: http://www.dcglug.org.uk/listfaq