Some things to do
1. Check codec's active on your hardware are active on your asterisk
2. Disable the firewall on your Asterisk box temporarily, it may be
blocking RTP ranges being used by hardware.
3. Check the asterisk logs for peculiar messages
/KB
On 06/12/2010 11:02, Niall O Broin wrote:
> On 6 Dec 2010, at 10:46, Barry O'Donovan wrote:
>>> Most likely NAT issues with the RTP streams in SIP.
>> I can't even get voice on a call from a hard phone to a softphone, both on my local network, so no NAT issues. Whereas I can make calls from softphone to softphone, or from softphones to/from the outside world, so the softphones aren't having NAT issues.
>>> Niall
>> --
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