[asterisk-users] SIP & NAT ...

[Anthony Francis]

Gordon Henderson wrote:
>
> 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
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do sip debug and then look again if still nothing then from linux do 
tcpdump -Avvv host <ip-address of problem device> and see if its getting 
blocked by iptables or not even reaching you. You should prolly show us 
what your sip.conf looks like and the dial command in use as well as the 
context it is in.