seeking optipoint sip help

apparently it wont work with internet sip providers, as it doesnt support stun servers. bugger.

That's not a big problem if you can configure your NAT router to forward to it the UDP ports used by SIP (5060) and by RTP (a range of ports usually configurable in the phone setup).

Enzo

its not the ports that is the problem. its authenticating with the sipgate server. i am unable to log on to my sipgate account, and therefore cannot make or receive calls.

OK, but the logging is achieved through "REGISTER" messages, and those travel inside UDP packets which, like any UDP packet, have source and destination port numbers (16-bit integers). By default, the SIP protocol uses the port number 5060, so, if your phone can't make use of a STUN server to gather information about the NAT and modify the content of its SIP messages to work around it, you may still be able to make it talk with the server by programming the NAT opportunely. What are brand and model of your NAT router?

Enzo

My knowledge of SIP etc is not very deep. my router is a us robotics sureconnect modem/router 9003.

I've never used that model, but from what I see at

it should be possible to configure it to make your SIP phone work. The most difficult thing is to guess the range of UDP ports used by your SIP phone for the RTP packets that convey the voice data. The precise port number used for each connection is communicated inside SIP packets, so you should tell the router to forward all the ports in the range to the phone, plus the port used for SIP which is normally 5060. If the documentation of your phone doesn't tell and you want to play it safe, you may always forward ALL the UDP "high" ports (between 1024 and

65535) to the phone; this will however prevent from working UDP-based applications (e.g., network games of filesharing programs) running on computers on the same LAN.

Also, I think that you have to program the router to allow all the outgoing UDP packets FROM the phone TO anywhere. This too is documented at

.

Good luck!

Enzo

thanks for that. ive had a go, but still no joy.

im not getting authentication failure message now, but i see four of these:

SIP/SDP Request: INVITE sip:10000@217.10.79.219:5060, with session description

10000 is the sipgate test number that i am dialling and 217.10.79.219 is the sipgate ip address and obviously 5060 is the port number. Obviously if i look into the messaging there is more stuff, but i cant work out whats going wrong.

Im looking at captures with ethereal, but while i sort of understand what i am looking at, i dont really know what to look for with regards to what the phone is trying to do.

Any more help greatly appreciated!!!

The "INVITE" message is the one sent by the caller to the called party, usually passing through a middleman (the "outbound proxy") belonging to the provider. In cases like yours the message is also sent to the provider's server because the device you are calling is supposed to have registered with it, and the server will either pass the "INVITE" to the called device or, more infrequently, reply to you asking to send it directly to the called device at the IP address so-and-so, much like a web server when it sends an "HTTP redirect" (that would be the so-called stateless proxying).

If the INVITE is repeated four times, probably your phone doesn't receive the "OK" from the server. This may be due to a number of reasons: the outgoing packet can't go through the router; the credentials (userID and password) that your phone uses to authenticate itself to the server are wrong; or the replies from the SIP server can't get in through the router and then arrive to the phone.

Even after this problem is solved, you might have a connecton but no audio in one or both directions: this is usually due to problems with the RTP packets that carry the voice data. Again, that could be due to the router blocking them, or to the UDP port mapping done in a way different from what phone and server have negotiated (that also depends on whether or not phone and server abide by the rules described by

... If your phone supports "symmetric RTP", do enable it: it may enhance the chances of getting the audio working.

If by now you are pulling your hair, take comfort in knowing that you are not, by any means, the only one... See e.g. the debate at

.

Unfortunatley Ethereal can only see the packets on the LAN side of the NAT, but can't tell you e.g. the UDP port numbers (both source and destination) on the external side.

This introduction, especially the sections 1.4 and 1.5, should give you an idea of how SIP works (or is supposed to work ;-) ):

This will help you making some sense of the packets sniffed by Ethereal. By the way, even before placing calls you should see the "REGISTER" transactions that your phone initiates in order to let the provider's server know its IP address and the fact that it's online. Until the REGISTER succeeds, there is little hope that other types of transactions (such as the INVITEs) may have better luck...

Cheers --

Enzo

Hi Rob,

I myself have done battle with the optipoint 400, and Oh what a battle!!! I am a bit of a novice but can share what basics I have found so far! I purchased the phone of ebay so thankfully didn't pay too much for it. The phone is available from hellodirect.com for above $300 in the states, and the American version uses a different firmware that looks better specified than the standard SIP firmware from siemens in .DE. However I have emailed both siemens in Germany and the States and my requests have been totally ignored!!. I cannot get the phone to log on to sipgate at all and I think I have tried everything in my power to try and sort the problem. I have also tried Gradwell, Gossiptel without success. Now the sort of good news.... I signed up for an account with voipfone.co.uk and to my surprise and joy!! I was able to receive and make calls, Oh the joy of hearing it ring!! After hours and days of wasted time trying to get it to work.

I have looked at the active nat sessions in my draytek routers configuration and the phone uses ports 5010 and 5011 for voice and port

5060 to communicate with.

It is not 100% as I have had a few calls drop audio and it failing to respond to the voipfone server fairly often when the call is cleared.

After extensive evaluation I have also discovered port 5060 is sometimes dropped from the NAT table, it comes back on and off during a call without affecting the call, but if you end the call while it has been dropped from the NAT table the phone reports "no server", sometimes it restores itself, other time only a reboot or disconnect / reconnect the LAN to the phone brings it back. Saying that I have been testing it just now and although port 5060 is shown in the table, after clearing down a test call the phone reports "no server" (it has just logged on itself after about 5 mins).

When it works the audio is first class, it really is a great phone and a joy to use, a shame about the siemens support!!.

I have the domain name set as voipfone.co.uk Registrar, server, gateway set to voipfone.co.uk OBP proxy nat.voipfone.co.uk Sip transport UDP Sip realm asterisk Sip user name (your voipfone number) Password (your password) Sip routing server Terminal number (your voipfone number)

It may be worth a try to see if it works for you.

Hope it helps!

John

thanks john! when i have the time i'll have a play. so on your router how exactly have you got NAT / port forwarding set up for the 3 port numbers you stated? (im a bit of a novice myself!)

Probably the best way is to try the phone in the DMZ of the router, that way it should work OK. I think my router is fairly VOIP friendly as I have found that it makes no difference if I forward the ports, put it in the DMZ or just leave it to sort its self out. You could also try forwarding UDP ports 5010, 5012, and 5060 to the IP address of the phone. John

i signed up with voipfone, but when i use your settings, i get this message: The submitted data contained errors:

• Invalid Terminal IP Address: voipfone.co.uk * Invalid Terminal IP Address: voipfone.co.uk * Invalid Terminal IP Address: voipfone.co.uk

im guessing for the registrar, server, and gateway addresses.

If i enter the ip address instead (212.187.162.78 then it takes it, but i get "no server" flashing on the display. I had it set for gateway when i tried to set up sipgate, so i changed it to that and it now looks ok. However, i still cant make any calls, so i guess i need to spend a bit more time. What version of f/w has your phone? mine is 2.3.14. i'll let you know how it goes.

Hi Rob

I certainly had the same errors as you initially but can't remember exactly how I solved the problem.

A few further thoughts,

I would reset the settings on the IP and routing page and then check you have the domain name field set to "voipfone.co.uk" I have a feeling this needed setting before the settings on the system information page were filled in, and saved.

I have had a play today to see if I could get the phone to behave itself after clearing down the call. It seems to make no difference whether the OBP setting is ticked or not, I currently have it un ticked and it makes no difference to the behaviour of the phone.

I also changed the OBP domain to voipfone.co.uk; again this doesn't seem to make any difference, so I guess the phone is not providing the correct OBP information. So guess this is probably why the phone keeps loosing the vopifone server.

Almost forgot, I am using the same firmware 2.3.14,

Hope this helps!

John

Fantastic! I can now make an outgoing call to a landline. I cant seem to call a sipgate phone (i have x-lite on my PC). i cant seem to receive a call though. I get either number incorrect from my mobile, or from my sipgate line i got a person unavailable message (possibly voicemail). My router doesn't have a DMZ. Maybe the port forwarding needs a bit more work.

Brilliant!!

The 056 number wont work from my virgin mobile, but OK from the landline.

I think the reason xlite wont call the siemens may be because they both use port 5060 for sip info, I seem to remember being unable to get xlite to work with an IP phone or xlite within the same LAN.I don't have xlite installed now.

At least it's heading the right way!!!

John

Sorry forgot to mention,

Forward UDP port "5060" Forward UDP ports "5010 to 5013" that way it includes port

5010,5011,5012,5013, as when the optipoint talks to another optipoint the next pair of ports are used. Forward these to the IP address of the phone.

John

ah! i was using a virgin mobile! trying to use 2 sip phones at once and it not working makes sense (cant see the wood for the trees!). of course they are trying to use the same port. doh! i'll have another play when i get a chance. thanks again.

just tried to call from a landline and an O2 mobile and i get the voipfone voicemail greeting. strange......

Wonder if the phone is not responding because the voipfone server can't see the phone behind the router properly.

I suppose when you make a call from the siemens it opens the ports through the router and so you are able to make a call.

I guess the phone sends out the register requests and the firewall/router/NAT passes these through to the server, and because the phone initiated the request, the response from the server is passed through the firewall to the phone and it is able to log on/ make calls.

If the server makes the request (for an incoming call) then maybe the firewall/NAT is unable to pass these correctly to the phone.

May be worth checking the configuration of the router as Enzo mentions earlier in the thread, I start to quickly get out of my depth when discussing the protocols of SIP port redirection etc!

As Enzo says, you would need to pass the ports (I mentioned earlier) from the IP of the phone to any destination in the routers configuration. May be worth doing a google for your router and SIP.

It can get so complicated, as life seems to be these days!

John