[asterisk-users] SIP & NAT ...

[Gordon Henderson]

So I thought I had SIP and NAT cracked a long time ago, but something's 
just happened that's sort of upset the cart )-:

I have an * box behind a NAT firewall. Nothing unusual there, this is 
something I've done many times - sip.conf has the correct

   nat=
   localnet=
   externip=

settings, the router has ports 5060-5069 and 10000-20000 forwarded to the 
internal IP address of the * box. (and 4569 for IAX, but we're just using 
SIP here)

The * server has a few internal (LAN) and external SIP phones, but also 
has 2 SIP connections to an external PSTN provider. I don't know what this 
is as I don't have any control or access to it, but both go to the same IP 
address with different account details (username/passwords)

Both these SIP -> external PSTN provider connections register OK on the * 
box, and outgoing calls placed over either connection works perfectly. 
Outgoing callerId (set by the external provider) works as expected. ) I 
have dialling prefixes for each 'line', nothing special there, that side 
of it all works as expected.

The problem is that only the last one in the sip.conf file actually 
accepts incoming calls when dialled from the PSTN side. (They have 
different PSTN phone numbers) If I swap their entries over in the sip.conf 
file, then the other one takes the calls.

When dialling the first number, nothing seems to get through to the * box 
at all - nothing on the console in verbose mode, nothing in the log-file.

The 2 SIP account setups are otherwise identical (generated by a web 
interface), just the usenrname & password differing, and the account name.

Anyone seen this before?

I'm wondering if it's an issue with the rotuter (Draytek 2800 ADSL), or is 
there an issue with 2 SIP channels to the same external IP address (port 
clash?) I've tried with & without bindport= settings in the sip.conf file 
too - doesn't make any difference.

Any clues appreciated!

Gordon