Koozali.org: home of the SME Server
Obsolete Releases => SME VoIP (Asterisk, SAIL etc) => Topic started by: Graham on July 15, 2008, 04:36:43 PM
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I'm trying to use the internet phone built into the Nokia N95 to connect to our VOIP server, I’ve can use X-Lite from a remote PC with no issues but when using the N95 it fails to connect.
Has anymore tried this before and got it working.
Regards,
Graham Spratt
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Forgot to say I can get the Nokia N95 to connect directly to a sipgate account.
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yes it works. Trouble is I can't find my notes on how to do it. The Asterisk side is dead easy - just set it up as general sip and set it as "remote".
The Nokia side is quite tricky if I remeber correctly, I'll see if I can find those notes...
Best
S
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here you go. This looks about right...
Profile Name: Asterisk SARK[or whatever you want]
Service profile: IETF
Default access point: [set up and put your Home access point here]
Public user name: sip:222@asterisk.server.com ["222"=your extension, + your ast server]
Use compression: No
Registration: When needed
Use security: No
Proxy server: [click]
Proxy server address: sip:asterisk.server.com [again your asterisk server]
Realm: asterisk [this stumped me for awhile - this is default, unless you've changed this]
User name: 222 [ie. your extension, same as above]
Password: 999 [your "secret" code defined from your extensions]
Allow loose routing: Yes
Transport type: UDP
Port: 5060
Registrar server: [click]
Registrar serv. addr: sip:Asterisk
Realm: None
User name: None
Password: None
Transport type: UDP
Port: 5060
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Can't seem to get it working:
What is Realm and where would it be set.
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realm is the SIP realm (asterisk uses "asterisk").
It should be one of the fields you fill out in the phone when you set up the SIP stack.
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Still can't get it work, below is my config
Profile Name: Asterisk
Service profile: IETF
Default access point: Network Name
Public user name: sip:4000@server.com
Use compression: No
Registration: When needed
Use security: No
Proxy server: [click]
Proxy server address: sip:server.com
Realm: asterisk
User name: 4000
Password:*******
Allow loose routing: Yes
Transport type: UDP
Port: 5060
Registrar server: [click]
Registrar serv. addr: sip:Asterisk
Realm: None
User name: None
Password: None
Transport type: UDP
Port: 5060
type=friend
username=4000
secret=********
mailbox=4000
host=dynamic
qualify=3000
canreinvite=no
context=internal
callerid="4000" <4000>
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw
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the proxy server address looks wrong. It should be your external domain name (if you have one) or your external IP address. Also I think your registrar server should be set the same. Finally, I think on our test phone (which I don't have here) we also set the registrar user and password to the extension and secret (same as for the proxy).
You appear not to have set the extension as external (looking at the sip.conf settings). You will need to set this for remote operations.
Kind Regards
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found the original post with our settings in...
http://forums.contribs.org/index.php?topic=38735.msg176265#msg176265
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Tried that also still not connecting, new settings below:
Profile Name: Asterisk
Service profile: IETF
Default access point: Network Name
Public user name: sip:4000@server.goes.here
Use compression: No
Registration: When needed
Use security: No
Proxy server: [click]
Proxy server address: sip:server.goes.here
Realm: asterisk
User name: 4000
Password:*******
Allow loose routing: Yes
Transport type: UDP
Port: 5060
Registrar server: [click]
Registrar serv. addr: sip:server.goes.here
Realm: asterisk
User name: 4000
Password:*******
Transport type: UDP
Port: 5060
asterisk settings
type=friend
username=4000
secret=****************
mailbox=4000
host=dynamic
qualify=3000
canreinvite=no
context=internal
callerid="4000" <4000>
pickupgroup=1
callgroup=1
disallow=all
allow=g729
nat=yes
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well at least it looks liike ours now.
What do you see at the asterisk console when you attempt to register the phone?
also, you can try running a packet trace on your asterisk box to see if the phone is attempting to communicate...
yum install wireshark
tethereal -R sip -i {eth0/eth1} -f "host {ip address of the phone}"
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Hi,
not sure if this is related to sprattgrahams issue..
I have a Nokia E65 and I assume the SIP phone works the same as a N95.
My E65 has been working happily, right up to this evening when I upgraded SAIL to 2.2.1-651
previously I was running -640
after the upgrade to -651 the phone will not connect, and I see the following on the asterisk console as it attempts
[Jul 22 19:42:36] WARNING[6869]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol } ¡6 } ¡½
[Jul 22 19:42:39] WARNING[6869]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol } ¡6 } ¡½
[Jul 22 19:42:45] WARNING[6869]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol } ¡6 } ¡½
[Jul 22 19:42:57] WARNING[6869]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol } ¡6 } ¡½
[Jul 22 19:43:13] WARNING[6869]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol } ¡6 } ¡½
[Jul 22 19:43:29] WARNING[6869]: chan_sip.c:6713 determine_firstline_parts: Bad request protocol } ¡6 } ¡½
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I haven't seen these messages before but Digium Bug tracker suggests it may have been a bug in Asterisk 1.4.19.
Which release of asterisk do you have?
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[root@l1nuxsvr ~]# rpm -q asterisk
warning: only V3 signatures can be verified, skipping V4 signature
asterisk-1.4.18.1-59.el4
Just updated to asterisk-1.4.21.1-64.el4
but still have the same errors on asterisk console and the Nokia SIP phone is "Unable to connect"
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A wireshark trace gives the following....
[root@l1nuxsvr ~]# tethereal -R sip -i eth0 -f "host 192.168.37.29"
Running as user "root" and group "root". This could be dangerous.
Capturing on eth0
0.000000 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
3.066796 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
9.138037 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
21.195069 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
37.274363 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
53.351243 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
69.414820 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
85.471164 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
101.544919 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
117.604870 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
133.669569 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
149.714726 192.168.37.29 -> 192.168.37.251 SIGCOMP/SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
12 packets captured
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Then I'm afraid I don't know. We don't have the issue here so I guess you will have to do a bit of detective work to find out what is causing the problem. You can back sail off to an earlier release (just remove the existing version with "rpm-e sail") or attempt to track which phone is causing the problem (we are only guessing it is the Nokia at this point). If you turn the Nokia off does the problem go away? Have you changed anything else?. Standard debugging I guess.
Kind Regards
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Hi Jeff,
in this case it is my bad !! (sorry to have wasted your time)
I looked at this thread yesterday and wondered if I could help, as my phone was working fine.
While looking at my phone settings, Thought (my 1st mistake) what happens if I turn compression on.
Hmm I guess I know now.. Yes setting compression to 'no' again sorted it.
Although.. I do get an error when it does register, and have always got this error..
Extension Changed 5019[extensions] new state Unavailable for Notify User Glen
-- Registered SIP '5019' at 192.168.37.29 port 5060 expires 180
-- Saved useragent "Nokia RM-208 2.0633.65.01" for peer 5019
Extension Changed 5019[extensions] new state Idle for Notify User Glen
[Jul 22 21:32:28] NOTICE[7343]: chan_sip.c:12669 handle_response_peerpoke: Peer '5019' is now Reachable. (151ms / 2000ms)
-- Got SIP response 400 "Bad Request" back from 192.168.37.29
So it seems the Nokia sends a Bad request? or is this request being interpretted wrong due to a SAIL setting for the extn ?
Here is a trace of registration.. don't know if it helps understand the bad request though,
[root@l1nuxsvr ~]# tethereal -R sip -i eth0 -f "host 192.168.37.29"
Running as user "root" and group "root". This could be dangerous.
Capturing on eth0
0.000000 192.168.37.29 -> 192.168.37.251 SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
0.000132 192.168.37.251 -> 192.168.37.29 SIP Status: 100 Trying (1 bindings)
0.000166 192.168.37.251 -> 192.168.37.29 SIP Status: 401 Unauthorized (0 bindings)
0.177820 192.168.37.29 -> 192.168.37.251 SIP Request: REGISTER sip:l1nuxsvr.routley.homeip.net;transport=UDP
0.178001 192.168.37.251 -> 192.168.37.29 SIP Status: 100 Trying (1 bindings)
0.229222 192.168.37.251 -> 192.168.37.29 SIP Request: OPTIONS sip:5019@192.168.37.29;transport=UDP
0.229285 192.168.37.251 -> 192.168.37.29 SIP Status: 200 OK (1 bindings)
0.229394 192.168.37.251 -> 192.168.37.29 SIP Request: NOTIFY sip:5019@192.168.37.29;transport=UDP
0.380837 192.168.37.29 -> 192.168.37.251 SIP Status: 200 OK
0.449795 192.168.37.29 -> 192.168.37.251 SIP Status: 400 Bad Request
10 packets captured
It doesn't cause a problem, just me not liking messages like that !!
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Our Siemens units do somethings similar. The phone is asking for something that asterisk can't/won't provide. I've never tracked it down completely because I found a report on the Digium site which said it was benign.
Glad you sorted the phoe out tho'
Best
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I got my Nokia E90 working perfectly fine through local wifi and through GPRS (3G/HSDPA), but i cannot seem to get it working through another ADSL connection with a sme server acting as gateway. con config on the fhone is as follows:
Profile name: Profile
Service profile: IETF
Default access point: {my wifi access point}
Public user name: sip:5000@mydynamichostname
Use compression: no
Registration: When needed
Use security: no
Proxy server
{all settings blank}
Registrar server
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Regstrar server address: sip:mydynamichostname
Realm: asterisk
User name: 5000
Password: {mypassword}
Transport type: Auto
Port: 5060
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Transportable SIP clients can never be very usable for traveling, using wireless zones, etc because of the nat/router unfriendly design of the SIP protocol.
The protocol that actually does work well for such use is the native Asterisk IAX2 protocol. For traveling I use a small PC with windows XP and Zoiper. My Nokia mobile SIP client works in something like 10 % of the hotspots. My Zoiper IAX2 client works in something like 98 % of all hotspots. (and I use the default IAX2 port and UDP 53 as alternative ports for logon.)
I whish it was a IAX2 client for Nokia, but I don't think there is. I think ther is one for Windows based mobiles, but I have never tried it. The Zoiper client of Slax Linux also works pretty well. I have tested the last one.
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Here is one of my old treads on this subject:
http://discussion.forum.nokia.com/forum/showthread.php?p=382599
I tested the Fring client at that time, but that version of Fring had terrible performace on my old E61, so it was not really usable.
I just tested Fring today, and it was qute much bether. I could connect to the sip server without a problem, and the sound quality had improved since last time tested. For reasons I don't know it worked quite ok to call out via sip but not to receive via sip.
Actually I think SIP via Fring is not really SIP from end to end, but it is sip to the Fring gateway and from the Fring gateway and to your mobile it is "Fring protocol". But I think the "Fring protocol and comunication concept" has bether performace trough nat routers (and firewalls) than the SIP protocol, so I will give it a try again .. :-) http://www.fring.com/
By the way, using your sip server with Fring will not require the configuration of the Nokia sip client but a quite more easy to set up "Fring sip client". (Because of reasons that is already mentioned.)