Sign in with
Sign up | Sign in
Your question

seeking optipoint sip help

Last response: in Networking
Share
Anonymous
May 31, 2005 7:02:31 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

I have a Siemens Optipoint 400 with SIP software.
I have managed to set it up to nearly work.
I am still unable to make or receive calls.

The ethereal trace shows:
SIP status : 407 Proxy Authentication Required
so it looks like its trying to log on with sipgate.

the sipgate softphone works fine from my PC.

Has anyone had any experience with this handset?
TIA.

More about : seeking optipoint sip

Anonymous
June 2, 2005 1:18:31 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

apparently it wont work with internet sip providers, as it doesnt
support stun servers.
bugger.
Anonymous
June 2, 2005 9:23:21 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Rob Walford wrote:
> apparently it wont work with internet sip providers, as it doesnt
> support stun servers.

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
Related resources
Anonymous
June 3, 2005 4:43:40 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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.
Anonymous
June 3, 2005 8:50:22 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Rob Walford wrote:
> 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
Anonymous
June 4, 2005 1:00:46 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

My knowledge of SIP etc is not very deep.
my router is a us robotics sureconnect modem/router 9003.
Anonymous
June 5, 2005 1:06:27 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Rob Walford wrote:
> 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
http://firewalling.com/usr/SureConnect9003-firewallallo... 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 http://firewalling.com/usr/SureConnect9003-firewallallo... .

Good luck!

Enzo
Anonymous
June 7, 2005 4:53:46 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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!!!
Anonymous
June 8, 2005 1:03:04 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Rob Walford wrote:
> 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

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).

> 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.

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
http://www.ietf.org/rfc/rfc3581.txt ... 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
http://www.isen.com/blog/2004/05/sip-was-good-idea-once... .

> 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.

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.

> Any more help greatly appreciated!!!

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

http://www.iptel.org/ser/doc/sip_intro/sip_introduction...

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
June 11, 2005 11:28:48 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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
Anonymous
June 11, 2005 3:11:36 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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!)
June 11, 2005 4:23:59 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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
Anonymous
June 11, 2005 4:52:09 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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.
Anonymous
June 12, 2005 3:02:55 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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
Anonymous
June 12, 2005 3:57:15 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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.
Anonymous
June 12, 2005 4:24:11 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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
Anonymous
June 12, 2005 5:43:06 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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
Anonymous
June 12, 2005 6:43:03 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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.
Anonymous
June 13, 2005 6:25:33 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

just tried to call from a landline and an O2 mobile and i get the
voipfone voicemail greeting.
strange......
Anonymous
June 13, 2005 1:04:19 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

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
Anonymous
June 13, 2005 5:22:14 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

i have set port forwarding, nat, firewall rules, anything else i can
think of and i can still only make outgoing calls.
If i remove all the programming, i can still make outgoing calls!
(i guess i have a stateful firewall)
I dont know what else to try. At least i can make outgoing calls,
though.
It must be doing something right to be able to do that.
I contacted voipfone, and they say that they cant even see my phone as
logged in.
I may eventually get a HandyTone-486 or similar, but if theissue is
with my router I may still have problems. Maybe i should eventually
get a sip-enabled router.
Anonymous
June 14, 2005 12:24:44 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

for ref, heres my f/w rules:

dir source dest srce port dest port pro tcp fl
action
any 0.0.0.0/32 192.168.1.8/32 1024-65535 1024-65535 UDP None
Allow
any 192.168.1.8/32 0.0.0.0/32 1024-65535 1024-65535 UDP None
Allow

heres my NAT entries:

Local Address Local Port Public Address Public Port Protocol
192.168.1.8 1024-65535 xxx.xxx.xxx.xxx 1024-65535 UDP
Anonymous
June 14, 2005 1:01:37 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

BTW, even with my firewall disabled, i cant receive incoming calls.
Anonymous
June 14, 2005 2:28:20 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

I am afraid I am not at all familiar with the way your router works and
am still trying to find out how mine works! (It's all a rather steep
learning curve!

I presume the public IP address in the NAT table is your IP address.

Not sure of the 0.0.0.0/32 in your previous post, the phone should
certainly communicate using the ports you have forwarded.

I also think rather than a firewall issue it is a problem with NAT
traversal (the phone not supporting stun or similar protocol)
I wonder is there a way you can put a destination address (i.e. IP of
Voipfone) in the settings (not even sure by mentioning this if its even
possible or desirable!)

Is it possible to see the active NAT sessions (my router is able to
show a real time table of active sessions) that way its possible to see
a little of what's going on.

I wonder if there may be some diagnostic software to sniff at the SIP
packets, not sure but think I have seen something that is able to look
at these mentioned in relation to SIP.

I rather like the look of the intertex range of routers, they have very
good SIP support but rather expensive! I would guess the phone should
work properly behind one of those; trouble is costs can easily runaway
with the various options available!

It seems the preferred ATA is the sipura range, I have seen a linksys
ATA around £ 45 on broadbandstuff.co.uk (I believe this is similar to
the sipura). The only trouble I found using an ATA was with echo on the
called parties end, this was very dependent on the phone being used. I
don't mind slight echo at my end (although a bit disconcerting) but
wanted the phone on the other end to sound as good as possible.

I have now settled for the snom 360 and find this is an excellent phone
but as with all this VOIP through the Internet it is not without its
latency and QOS issues at times.

If the firmware was upgraded then I am sure they could easily add the
features needed to make it work properly! The American version has a
higher version number, but I have failed to find a means of downloading
it.

Any one out there have any thoughts of how the firmware can be
obtained, I have tried emailing Siemens without response?

John
Anonymous
June 14, 2005 3:57:10 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

0.0.0.0/32 should mean "any ip address" on my router.
i am currently in email communication with voicefone.
he asked me to try nat.voicefone.co.uk:5065 as the outbound proxy, but
he obp domain entry in the optipoint would not take it.
i have made one call, and was certainly impressed by the quality.
It seems odd that you can receive incoming and i cant, thats why i
think its a router/firewall/nat issue somewhere.
i'll let you know if i get anywhere with voicefone support.
i asked about disabling the voicemail, as to test i have to call and it
costs me every time, but that option is coming soon apparently.
I d/l the handytone 486 user guide. What i like about it is that you
can select to dial voip or landline, although it has a built in router
with NAT etc and i am not sure how it would sit behind my existing
router. i'll have a look at the sipura as well.
Anonymous
June 15, 2005 2:17:32 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

I think the OBP setting port number is internally set; I have no idea
what it is.
The phone is very restrictive of what settings you can make, i.e.
setting the obp to a specific port number, etc.
If only it was possible to get hold of the later American firmware for
it!
John
Anonymous
June 19, 2005 9:23:56 PM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Rob Walford wrote:
> BTW, even with my firewall disabled, i cant receive incoming calls.

Disabling the firewall may still leave the NAT issues to be tackled.

What happens if you disconnect the IP phone and use an Xten or other
softphone?

And what if you use another service provider such as FWD? At
http://www.freeworlddialup.com/content/view/full/274/ you may find a
step-by-step guide, including detailed configuration instructions for
the X-Lite. Once the X-Lite works with FWD, you may assume that your
router has been tamed, and have the hardware IP phone work with FWD.
Finally, have the IP phone work with your provider of choice. Divide
and rule! :-)

Enzo
Anonymous
June 21, 2005 3:39:44 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

x-lite has always worked fine with sipgate, but i didn't set it up.
Sipgate automatically installs x-lite with all the correct settings.
I suppose that means that the issue should lie with the phone, but its
slightly frustrating if John managed to get it working.
Anonymous
June 28, 2005 11:22:33 AM

Archived from groups: comp.dcom.voice-over-ip (More info?)

Just a quick update.
I have also tried the phone with a linksys hs200 router and it seems to
work ok.
I certainly think it is a NAT problem with your router.
If I think of any thing else I will let you know.
John
!