This is my first Asterisk project and I am having a problem regarding SIP.
I have set up an Asterisk server (version 1.2.4) and I need to connect to a service provider through SIP (they don’t support any other protocol and they aren’t using Asterisk) to get PSTN connectivity.
There is no NAT being used on my side but on the provider side NAT is being used.
I can make calls from the PSTN to one of the VoIP phones, so the conversion from analogue to digital works fine, but I can’t make outgoing calls. When I try to call out the phone rings but it just times out. Is this because of NAT?
Could anyone please give me some advise about how to solve this or where I should look? I have already read the Wiki and numerous posts on this forum and they have been, and still are, very helpfull, but I need some help on this one.
I can’t post any output from the CLI or configuration details right now, but I’ll do so tomorrow, sorry about that.
Thanks in advance for any suggestions.