SIP Time OUT using Private IP???

[dean-ny]

Hello,

I have a trixbox with a handful of extenions both on and off my private LAN. Everything inside my network works fine. I also have a couple extenions off my network (publi internet) working fine with no change special configuration. One is an X-Lite and one is a 286. However I also have a couple x-lites and one 286 that are all in different locations that get time-out errors. They all are behind routers/nat even those that are working. The ones that are not working however register in tb as the private address. The ones that work properly are registering as the public address on the extension owner's network. For example the 286 is showing up in tb as 192.168.0.2 which is the LOCAL address of that device on that network. Any suggestions on why some networks and some don't. Thanks in advance for any help.

[mberlant]

By "time-out errors", do you mean that these clients never Register or that they lose Registration from time to time?

Your comment about the remote LAN local address showing on your Asterisk indicates that these remote clients do not have STUN configured correctly (most likely) and/or they are behind Symmetric NAT routers (least likely) and/or they are behind NAT routers that are not SIP aware and therefore do not cope with the client not having STUN working.

[dean-ny]

Initially my answer would be that the SIP or X-Lite client would timeout while trying to register. That is still the case for the X-Lite clients, and currently the case with the SIP. However, the SIP (ATA286), managed to register overnight while just sitting. The really wierd thing is that it registered with the correct public IP this time. That is at least how it showed up on the * info page in trixbox. Once this happened, the SIP could dial another extension but there was no audio incomming or outgoing audio. And, when you dial that sip from another extension * would regognize it as a valid registered extension and allow you to dial it. However after 5 rings it would go to voicemail, and the phone hooked to that sip would never ring. As a troubleshooting measure, I forward 5060 -5063 and 10,000 - 35000 to that IP on the Net Gear RP614V2 router. After rebooting everything, nothing worked. The red light blinks on the Sip and no registration shows up in *. It hasn't registered since doing the port fowarding. My next step will be to remove that in the router and see if I can at least get back to where I was.

Regarding STUN, there is no STUN server in my * (TB) box. Forgive me, but * is pretty new to me. But, I was advised that I didn't need STUN. And I was advised that I didn't need port fowarding on the remote stations. Ironically enough, I have remote X-lite clients that are working perfectly without port fowarding and without the stun server. However, both sites that don't work, do work with FWD's server and that is set up with STUN but no port fowarding.

Now, what is the best recomendation for getting STUN on my box if that will solve the problem? Is there an alternate way to configure the clients without STUN? Why would some clients work without both, or are they just lucky? The clients working are behind a Linksys router.


Oh, and everything is NAT'ed.

Thanks for the help.

[mberlant]

As you read through the previous postings here, you will learn that STUN is a service to help clients traverse an outbound NAT boundary, and is not used by a server. Details of this process are described in other threads in this forum.

Asterisk is a server with client capabilities. To extensions Registering on it, it is a server. To VoIP services you subscribe to, it is a client. Since Asterisk is a server, it must be exposed to the public internet in order to receive Registrations and ad hoc inbound calls. That is why you forward Ports UDP 4569, 5060 and 10000-20000 to Asterisk. Since these ports are forwarded to Asterisk on account of its role as a server, it would be redundant to equip Asterisk with a STUN client, so none is included.

All of your remote clients need to traverse the NAT router at that client's location.

Some routers are SIP aware, knowing how to translate the headers of the actual SIP packets to masquerade as the client in the face of the server. In this case, nothing needs to be done to traverse this NAT boundary.

Some routers are cooperative with SIP clients, which use NAT discovery techniques to learn the public IP information directly from the router. In this case, STUN is not required.

A very few routers are of the Symmetric NAT variety. These routers are wholly uncooperative with SIP, so internal IP addresses, port assignments and port forwarding assignments must be manually managed for each and all SIP clients residing behind that router.

Most routers don't fall into any of these other categories, and benefit from STUN. The client then must be configured to contact a STUN server to learn its public IP information, as assigned by the client's NAT router, which the client then includes in its Register request to the SIP server.

The new eyeBeam-like version of X-Lite comes with its STUN client disabled. You need to enable it and choose a sever to use. Any server will do; I tend to use stun.fwdnet.net:3478, but you may choose any one that works.

As a self professed newbie, you have come to the right place to learn about Asterisk. There's a wealth of information in the threads here. You just need to read.

[dean-ny]

I sincerely thank you for all of that informaton. That really does clarify a lot of things. It is really nice to have knowable people help us become knowelegble.

I was not at all aware you can you a third party STUN. My trixbox and router were professionaly set up so based on your explaination, I suspect that STUN is the answer. If I am understanding you, I can simply add FWD's Stun Server to the SIP device, remove the port fowarding, and it should work?

Thanks again :-)

[mberlant]

That's how I do it. I've written here many times that I use the Voxilla Wizard for FWD w/STUN to do 99% of the configuration for a Linksys/Sipura device and then go into the web page to change "fwd.pulver.com" to the FQDN of the Asterisk.

[dean-ny]

Well the good news is that with STUN added to both x-lite and that ATA-286 both on different networks, I can now make the phones ring. Specifically, I can call X-Lite on Network 1 and I can call the ATA on network two and the phones ring. Likewise, I can call from X-Lite and the ATA to any extension inside my network. The problem is, that there is no audio. I believe the problem from reading other posts is protocal related. Trixbox is set up to use G711U. Specific settings are below. The device is set for PCMU.

The other wierd things are that when you call from and external extenstion the phone will stop ringing and you get silence on the calling party side. However if you call from an internal extension to an external one, you the phone never stops ringing (outbound ring signal) on the callers side even when the personal called goes off hook.

Making progress, but still have a way to go. I'll continue to try to find similar posts but if you have any other ideas.....

Thanks.

dtmfmode
canreinvite NO
context from-internal
host dynamic
type friend
nat yes
port 5060
qualify no
callgroup
pickupgroup
disallow
allow
dial sip/2504
accountcode
mailbox

[mberlant]

The parameters you listed are from where in sip.conf? They look like they are from an extension definition, except for the "port 5060" line, which can hurt you if you aren't micromanaging port assignments. If this is from an extension definition, try changing it to port=dynamic.

[dean-ny]

It is directly from the extensions screen in trixbox. I will try the dynamic. I'll try anthing at this point :-)

[dean-ny]

OK, along with your advise and a router reboot everything is working fine now.