asterisk users - Jun 2007 - SIP & NAT ...

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

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
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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.

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?

Tom

On Fri, 1 Jun 2007, 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.
So I have now had to time to get to the bottom of this and it seems to be 
nothing to do with NAT.

I even found some pages on the WiKi that were useful: (Although not under 
any heading I'd have thought of!)

http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer

or at least talked about others having the same problem and presents some 
solutions one of which I managed to adapt for my situation.

I'm not sure if I'd call this a bug, feature or just a PITA )-:

Incoming SIP calls (from registered hosts, when we're a peer?) start from 
the bottom of the sip.conf file and work upwards. However the bottom entry 
would appear to be the one that's used to authenticate all of this type of 
incoming SIP calls, the others aren't.

So I created a new peer entry at the bottom of the file, insecure=invite, 
and a new context to send incoming calls to and it seems to work. The 
far-end is OpenSER in-case anyone knows of that having issues.

And while it works, I don't think it's too satisfactory a solution, (and
I
can't work out how to put this into a GUI yet!) but there you go.

Gordon