| Author |
Message |
Rob Walford
Guest
|
 Posted:
Wed Jun 01, 2005 12:20 am Post subject:
seeking optipoint sip help |
|
|
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. |
|
| Back to top |
|
 |
Rob Walford
Guest
|
| Posted:
Thu Jun 02, 2005 4:20 pm Post subject:
Re: seeking optipoint sip help |
|
|
apparently it wont work with internet sip providers, as it doesnt
support stun servers.
bugger. |
|
| Back to top |
|
|
Enzo Michelangeli
Guest
|
| Posted:
Fri Jun 03, 2005 5:23 am Post subject:
Re: seeking optipoint sip help |
|
|
Rob Walford wrote:
| Quote: | 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 |
|
| Back to top |
|
|
Enzo Michelangeli
Guest
|
| Posted:
Sat Jun 04, 2005 12:20 am Post subject:
Re: seeking optipoint sip help |
|
|
Rob Walford wrote:
| Quote: | 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 |
|
| Back to top |
|
|
Rob Walford
Guest
|
| Posted:
Sat Jun 04, 2005 12:20 am Post subject:
Re: seeking optipoint sip help |
|
|
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. |
|
| Back to top |
|
|
Rob Walford
Guest
|
| Posted:
Sat Jun 04, 2005 4:20 pm Post subject:
Re: seeking optipoint sip help |
|
|
My knowledge of SIP etc is not very deep.
my router is a us robotics sureconnect modem/router 9003. |
|
| Back to top |
|
|
Enzo Michelangeli
Guest
|
| Posted:
Sun Jun 05, 2005 8:20 am Post subject:
Re: seeking optipoint sip help |
|
|
Rob Walford wrote:
| Quote: | 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-firewallallow.htm 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-firewallallow.htm .
Good luck!
Enzo |
|
| Back to top |
|
|
Rob Walford
Guest
|
| Posted:
Wed Jun 08, 2005 12:20 am Post subject:
Re: seeking optipoint sip help |
|
|
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!!! |
|
| Back to top |
|
|
Enzo Michelangeli
Guest
|
| Posted:
Wed Jun 08, 2005 8:20 am Post subject:
Re: seeking optipoint sip help |
|
|
Rob Walford wrote:
| Quote: | 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).
| Quote: | 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.html .
| Quote: | 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.
| Quote: | 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.html
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 |
|
| Back to top |
|
|
John
Guest
|
| Posted:
Sat Jun 11, 2005 4:20 pm Post subject:
Re: seeking optipoint sip help |
|
|
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 |
|
| Back to top |
|
|
Rob Walford
Guest
|
| Posted:
Sat Jun 11, 2005 11:11 pm Post subject:
Re: seeking optipoint sip help |
|
|
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!) |
|
| Back to top |
|
|
John
Guest
|
| Posted:
Sun Jun 12, 2005 12:21 am Post subject:
Re: seeking optipoint sip help |
|
|
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 |
|
| Back to top |
|
|
Rob Walford
Guest
|
| Posted:
Sun Jun 12, 2005 12:21 am Post subject:
Re: seeking optipoint sip help |
|
|
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. |
|
| Back to top |
|
|
RJHN *1JOD
Guest
|
| Posted:
Sun Jun 12, 2005 11:02 pm Post subject:
Re: seeking optipoint sip help |
|
|
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 |
|
| Back to top |
|
|
Rob Walford
Guest
|
| Posted:
Sun Jun 12, 2005 11:57 pm Post subject:
Re: seeking optipoint sip help |
|
|
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. |
|
| Back to top |
|
|
|
|
|
|
FAQ
Memberlist
Register
Profile
Log in to check your private messages
Log in