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Hi folks,I'm having a bit of an issue with Asterisk and SIP calls from an external server and I thought I'd ask the knowledgeable folks here if they might be able to shed some light on it?
Basically the setup is like this:- Asterisk server behind NAT with a couple of SIP phones on the network, calls to Asterisk (Voicemail) and between the phones are fine. Outgoing calls over the analogue cards work (not great quality, but they work).
- External SIP number from Sipgate.co.uk for testing purposes. Asterisk connects to Sipgate's servers ok.
If anyone calls the external SIP number (which terminates to a local 01803 number) we can hear whoever answers the call but they can't hear the caller (I'm presuming they're just getting silence).
I've enabled ports in NAT so that port 5060 (UDP) is forwarded to the Asterisk server and that another range of ports are forwarded too a per a guide on 3CX (couldn't find anything specific to Asterisk, but I understand that 3CX uses SIP too).
What I wasn't sure about is how the calls are routed.Once a call is answered does the Sipgate server try and make a direct connection to the IP Phone that answers, or does it still route the call through Asterisk and then on to the IP Phone?
Basically I'm wondering if we need to open ports in NAT for every IP phone or just Asterisk itself?
Rob -- The Mailing List for the Devon & Cornwall LUG http://mailman.dclug.org.uk/listinfo/list FAQ: http://www.dcglug.org.uk/listfaq