[asterisk-users] SIP & NAT ...

[Gordon Henderson]

On Fri, 1 Jun 2007, Tom Rymes wrote:

> On Jun 1, 2007, at 9:45 AM, Gordon Henderson wrote:
>
> [snip]
>
>> 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.
>
> [snip]
>
> I may be mistaken here, but don't you need to use different ports for each 
> line? ie: Port 5060 for line 1 and 5061 for line 2?

Well, this is something I'm not 100% sure about. Sip.conf has port= and 
bindport= parameters, and as far as I can make out port= means use this 
port to connect to the remote server, and bindport= means listen to this 
point for incoming calls.

port= didn't work. Not surprisingly because the remote server is listening 
on 5060 only.

bindport= (and I tried differnet ports for each account - 5062 and 5064) 
didn't seem to make any difference. Whether this was being absorbed by the 
NAT functions on the Draytek, I don't yet know. Will do more experiments 
over the weekend.

Thanks,

Gordon