E61 SIP client

Hello!
I'm trying to setup my E61 as SIP Phone.
My private SIP server is accessible from Internet (it has public IP), my phone has public IP (with GPRS connection).
SIP client on E61 connects to my SIP server and sends REGISTER
request without username and password, SIP server answers with
"401 Unauthorized", but phone continues to send REGISTER requests without username and password (while I have configured in SIP profile settings both username and password)
If I use another SIP client (for example Linksys SPA922) it sends first REGISTER without username and password too, but after receiving "Unauthorized" it sends REGISTER with username and password.
I have checked E61 firmware with Nokia Software Updater and got message that I have latest version.

Ok, after a day work, it starts fine
Here is my working profile
Service profile: IETF
public user name: sip:[email protected]
use compression: no
registration: when needed
use security: no
Proxy:
server address: sip:192.168.0.1
realm: avaya.pbxlaba
username: 601
allow loose routing: yes
transport type: udp
port: 5060
Registrar:
serv addr: sip:192.168.0.1
realm: avaya.pbxlaba
username: 601
transport type: udp
port: 5060

Similar Messages

E61 - SIP/VoIP

I've just bought a Nokia E61 - mainly for its SIP functionality.
I have entered all my SIP settings, and the phone just doesn't register with the SIP server. I can't even see the attempt to register in the server logs.
What makes me wonder is also that I don't get to choose the connection type for the SIP connection (e.g. my wireless LAN) etc.
I called Nokia support and they tell me that they don't support the SIP client. Great. That's what I bought the phone for ...
Has anyone managed to get the SIP client working? I'm using an asterisk server.
Kind Regards,
Christian

Although I have not verified this myself (I have an E60 not an E61) a poster over on my-symbian.com reports that the new firmware (version 3) for the E61 supports STUN. If this is true you should be able to register and use Vonage. Vonage will not let you use your regular VoIP service (the one that goes through their router). You need to get a "softphone" for an addition $10/month. Then your settings will be as follows:
Profile name: Vonage
Service profile: IETF
Default access point: your wireless router (NOTE: you need to specify an access point, it will not ask for one)
Public user name:sip:[email protected]
Use compression: no
Registration: When needed or Always
Use security: no
Proxy server:
Proxy server address: none (leave this blank)
Realm: 216.115.20.41
User name: your softphone number (including the 1)
Password: your softphone password
Allow loose routing: yes
Transport type: UDP
Port: 5060
Registrar server:
Registrar serv. addr.: sip:sphone.vopr.vonage.net
Realm: sphone.vopr.vonage.net (or 216.115.20.41)
User name: your softphone number (including the 1)
Password: your softphone password
Transport type: UDP
Port: 5061 (or 5060)
I was able to use this to register and make 30 second calls but without STUN, I could not maintain the connection.
MarkMessage Edited by mjlaris on 20-Dec-2006
01:56 AM

Why does a SIP Client in background not answer calls?

Using any of the SIP clients on IOS7.1 it appears that I have about 10 -15 mins before a SIP client in the background becomes unresponsive. Any ideas why and what can I do about it?

Since this is an app-related issue, you may get better results solving your problem by contacting the app developer, or searching their help/FAQ pages.

VoIP SIP Client

Now I know I'm not the only one seeking the ability to write a snazzy little Flash Application for SIP access. I am, however, the only one willing to start a neat little open source project to help, people like me, use SIP to its full potential.
So here it goes, I'd like (and hope) to write a SIP client for two projects. The first is a for-profit endeavor for the company I work for, and the second is a not-really-for-profit webservice I own that would have its customers benefit from on-site VoIP call. Since they both require the same thing, I'm willing to open it up as an open source project to allow it to continue growing.
The idea is this:
A list of contacts (unassociated with the SIP client) have phone numbers or VoIP extensions. Click on the call button will activate a snippet of javascript that will communicate with the Flash SIP Client, sending instructions to dial. On the bottom of the screen the flash client will begin dialing, or ask for permission access cam and mic then dial. The sip client must also be able to receive inbound calls being projected to it from my VoIP server. Do all the things needed for a SIP client to do, such as hold, conference, mute, answer, and hangup calls.
So I'm thinking the Flash Client will be a thin, virtually no interface, that will communicate with the onboard javascript. This thin-layer of flash would be invisible, to everyone except a developer maybe looking at trace information. Javascript would instruct the flash client to connect to SIP, establish calls, hangup, etc. Flash would also send instructions from the VoIP server to javascript such as incoming calls, text, sign off, messages... etc.
So what I'm asking you as the community here at adobe for is somehelp. I've been Googling and found not to much helpful in this area.
I know flash can communicate to javascript. I know Javascript can communicate to flash (Yahoo does it for their IM client). I know Flash can communicate with VoIP servers via sip. What I don't know is where to start writing this client. I herd that AIR has an API for such things, that maybe even flash has a SIP/VoIP API, where is it? If I have all this information I'll start a nice open source project on sourceforge, github, or something like that where I hope to get input and offer the very thing I'm looking for to people all over the net. Expand VoIP capabilities so we can truly see inexpensive solutions popping up over the net. With the advent of VoIP integrated telephony should not be an expensive effort.
P.S. - Nothing says that when you sign in that the Flash API and Javascript components don't communicate with each other through another window to keep the SIP client connected. Also nothing says that when the main windows closed it doesn't disconnect the sip client and the user goes on his/her merry way... This IS possible, I just need to put the larger peices together.
Let me know!

I am definitively interested. Will you contact me at radoslav <At> everestkc.net.
What about this: http://flashphoner.com/
Is this totaly open source or you need to pay to be able to check out the code?
Rad

SIP Client for Symbian (Nokia N91)

Anybody have a SIP Client for Nokia N91 (Symbian OS)or knows dates of releases, links, any thing?
Thanks
Juan

Did you get a SIP client for the N91? if so where from.

Vmobile SIP Client

I use an Asterisk open-source SIP server in my home office as mu business telephone system. Works great. I downloaded the Vmobile SIP client for my 9930 Bold so I could use it as an extension of the Asterisk. Two issues: I get a DNS error when attempting to log in to my home wifi network even though the domain name is recognized by other apps. When I tried to switch to the 3G cellular network, I am prompted for an "APN name, APN username and APN password". I cannot find this info anywhere.
I did try to get to the blackberry.vmobile.eu webpage, but it appears to be down tonight.
Anyone have any suggestions>

sorry just a question then maybe a worthwhile reply.
where the heck do you get this "vmobile" client from??

SIP client for C3-00 ?

Is ther any SIP client for C3-00.
I've heard of fring, but it does not seems to be compatible with C3-00.
I've herad of x-lite, but not compatible either with C3-00...
any advice ?
TIA
Alain

There have been few threads around this,
/t5/Cseries/Can-I-do-voip-on-the-c3-00-wifi-s40-sip/m-p/761253 and http://discussion.forum.nokia.com/forum/showthread.php?209961-Can-I-do-voip-on-the-c3-00-(wifi)-s40-... it seems like VoIP is not possible on C3-00.
If you are looking for VoIP capable Nokia device then perhaps this list in Forum Nokia could help you further, http://wiki.forum.nokia.com/index.php/VoIP_support_in_Nokia_devices#Support_in_Series_40_devices
Hope it helps

Why is E61i SIP client pinging strange IPs?

I reflashed my E61i (using NOKIA's software updater). Then installed Trend Micro's Web Security and set the firewall to high (disallowing outgoing packets except those specifically allowed eg http)
Wait a day. Check firewall logs and they're clean.
Install Handy Clock.
Wait another day, check firewall logs and they're clean.
Change firewall settings to allow SIP client to connect to SIP provider only. Setup SIP client for my VOIP provider.
E61i starts pinging IPs in china, spain, argentina, India, UK, Comcast etc at random intervals, sometimes as short as 2 minutes. Some of the IPs appear to be ADSL IPs.
Anyone has any explanation why NOKIA's SIP client should be pinging all these IPs? Is there a vulnerability in the SIP client?
Can anyone recommend a firewall which can tell you which process or application is trying to connect to the internet?
Don

Please find :
C:\>ipconfig
Windows IP Configuration
Ethernet adapter Local Area Connection:
   Media State . . . . . . . . . . . : Media disconnected
   Connection-specific DNS Suffix . :
Wireless LAN adapter Wireless Network Connection:
   Connection-specific DNS Suffix . : srhouse.com
   IPv4 Address. . . . . . . . . . . : 172.21.155.24
   Subnet Mask . . . . . . . . . . . : 255.255.255.0
   Default Gateway . . . . . . . . . : 172.21.155.1
C:\>

Nokia 5800 VOIP SIP client that works!

I have just installed the V Phone VoIP SIP client onto my UK Nokia 5800, and it works very well, "straight out of the box". I have it set up to use my sipgate.co.uk account, and it registers via WiFi immediately, with good clear call quality. Sometimes there is a bit of echo, but generally the call is clean.
So here's a Nokia 5800 SIP VoIP application that does work - I thought that such a thing didn't exist. V Phone are an Australian outfit, found here: http://www.thevphone.biz/Products.html. I bought their Premium version for about £5.50 equivalent of Australian dollars, so it isn't expensive either.
Recommended (and I don't have anything to do with them, if you were wondering).
Tom

Does it really work? My operator requires proxy. Is there an option in the settings to enter proxy server? I went on their website and saw there is only registrar, user and password. Also, when I try to purchase Premium Edition with G729 codec, I get an option to purchase V Phone - s60 - Standard Edition for 8.95 dollars. The price is really good but I wonder if it really works.

N80i SIP client problem "unable to connect"

I'm having a problem with my N80i. When i use my phone in my WLAN for browsing the internet everything works fine. The problem appears when I try to use de SIP client, i always get "unable to connect to network". I've configured a Gizmo account,but can't connect. I've also configure an asterisk SIP account on my local network (in order to avoid any routing/port blocking/fw issue) and the same thing happens. In fact I've activated a sniffer in my linux box and I don't see any packet coming from the phone.
As far as I've investigated it seems like the problem occurs inside the phone (no ip packets come from the N80).
I wonder if it might be a firmware block.
Any help will be really appreaciate

I want to share with you my experience, just in case someone is having the same problem I had.
I could solve the problem. Here is the config I use in the SIP settings:
Profile name: {whatever U want}
Service profile: IETF
Default access point: {a WLAN access previously define as access point from the WLAN wizzard}
Public user name: sip:{user}@{Server IP or name}
Use compresion: no
Registration: always on
Use security: no
Proxy Server:
Proxy server address: sip:{server ip or name}
Realm: asterisk (in case u are using asterisk or the same name defined in proxy server address)
User name: {user}
Password: {password}
Allow loose routing: Yes
Transport Type: UDP
Port: 5060
Registar Server: {same settings like proxy server}
I've to mention that while i was trying to make it work, i downloaded GizmoVoip (without success), but when I tried Truphone (www.truphone.com) it did work. So what i did was to copy exactly the same profile (SIP Settings->Options->Add new->Use an existing profile->Truphone-home) and with that i created a new profile. After that I changed the config to match my asterisk and....IT'VE WORKED!!!!
So, the conclution: I think the problem i was experiecing was due to a missconfig in the "proxy server address" or "Public user name", I'm not sure if I was putting the "sip:" at the beginning (i made to many tests that i can't remember). If that was the mistake then it seems like the Nokia N80 was not even trying to connect to the server and that was the reason why I was not seeing any packet coming from the phone with the sniffer.
I hope this info will be useful for everyone.
Martin

E66 SIP Client

Hello,
I am having troubles with the voip client of my E66.
First when making a phone call, the speedtouch 780 reboots after around 10 minutes talking. I'm using the xs4all network in the netherlands. Other phones from Nokia (E60 and E65), and the internal SIP client of the speedtouch and a network attached Linksys SIP client behave well. I have had this problem on multiple networks.
Second when calling my E66, it reports directly a missed call. The caller gets the message the E66 is not available. I am not having this problem with the E60 and E65.
Saco

lets see:
Any firewall settings in the way, the usual SIP port for messaging is 5060, is this open?
Are u using WLAN or 3G or GPRS.
If u r using a WLAN connection then this is a bug in the phone just like in the E65 some time ago. You can't connect to SIP if the router uses static IP address. So you can only connect to SIP when the router uses DHCP and the phone get a dynamic IP address. I hope it will be patched soon...
Articles posted courtesy engadget
keep us updated about the progress.... if u like wat I have to offer then click on khudos.

SIP client on RIM BBs

Hi All,
I could not find any satisfactory info about SIP on BB.
My questions are:
1) Is there any sip client possible to work on RIM BBs?
2) If yes, which models of RIM BBs support SIP client work on it?
3)Does any RIM BB have a sip client on it?
Thanks in advance
[email protected]

For now there is no SIP client for Blackberry available.
There are a lot of sites which declare such products, but all of them are in "Coming soon" state.

SIP client and delayed SDP offer-answer

Hi,
I'm trying to connect the E51 SIP client to the company PBX.
Calling out works fine, but I can't receive calls from the PBX.
It seems that the source of the problem is that the PBX is sending INVITEs without an SDP offer.
Do you know if the E51 SIP client supports SDP delayed offer-answer?
Thankes,
Dan

Well,
The reason is not the delayed offer-answer. The E51 accepts SIP calls from X-Lite. But, not from the PBX (even when SDP is present in initial INVITE). For the PBX it responds with: 480 Temporarily Not Available (see below)
Which is quite interesting...
Maybe it is because all those additional SIP headers.
Any thoughts?
Dan
Oct 29 17:53:58 2008 : [Send Request ]
{connection: host=135.64.17.239 port=5060 protocol=UDP}
INVITE sip:[email protected];transport=UDP SIP/2.0
Call-ID: 012bdb8aba6dd1dd349c602800
CSeq: 1 INVITE
From: "Dan Gluskin" ;tag=012bdb8aba6dd1dc349c602800
Record-Route: ,
To: "37400"
Via: SIP/2.0/UDP 149.49.130.131:5060;branch=z9hG4bK0303031313132323235f4.0,SIP/2.0/TLS 149.49.130.131:6001;psrrposn=2;received=149.49.130.131;branch=z9hG4bK012bdb8aba6dd1de349c602800
Content-Length: 304
Content-Type: application/sdp
Contact: "Dan Gluskin"
Max-Forwards: 69
User-Agent: Avaya CM/R013w.01.2.024.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS
History-Info: ;index=1
History-Info: "37400" ;index=1.1
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: ;avaya-cm-alert-type=internal
Min-SE: 1200
Session-Expires: 1200;refresher=uac
P-Asserted-Identity: "Dan Gluskin"
v=0
o=- 1 1 IN IP4 149.49.130.131
s=-
c=IN IP4 149.49.130.132
t=0 0
m=audio 2376 RTP/AVP 0 8 18 110 4 127
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 G726-32/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:127 telephone-event/8000
Oct 29 17:53:58 2008 : [Recv Response ]
{connection: host=135.64.17.239 port=5060 protocol=UDP}
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 149.49.130.131:5060;branch=z9hG4bK0303031313132323235f4.0,SIP/2.0/TLS 149.49.130.131:6001;psrrposn=2;received=149.49.130.131;branch=z9hG4bK012bdb8aba6dd1de349c602800
To: "37400"
From: "Dan Gluskin" ;tag=012bdb8aba6dd1dc349c602800
Call-ID: 012bdb8aba6dd1dd349c602800
CSeq: 1 INVITE
Content-Length: 0
Oct 29 17:53:58 2008 : [Recv Response ]
{connection: host=135.64.17.239 port=5060 protocol=UDP}
SIP/2.0 480 Temporarily Not Available
Via: SIP/2.0/UDP 149.49.130.131:5060;branch=z9hG4bK0303031313132323235f4.0,SIP/2.0/TLS 149.49.130.131:6001;psrrposn=2;received=149.49.130.131;branch=z9hG4bK012bdb8aba6dd1de349c602800
To: "37400" ;tag=f9srkqi8iphc6t5c9eru
From: "Dan Gluskin" ;tag=012bdb8aba6dd1dc349c602800
Call-ID: 012bdb8aba6dd1dd349c602800
CSeq: 1 INVITE
Content-Length: 0

E61 SIP realm bug !

Hello,
[My E61 use the latest firmware from july 2006]
There is an important bug with the E61 SIP.
Its impossible to create 2 differents accounts with the same realm (domain) name !?!?
Exemple, i have two accounts with realm *freephonie.net*, but E61 dont accept the second account.
Perhaps, one day..., this #!:#=# !!! bug will be removed.
But i must to find a solution, now.
So i would be glad to know the name of the file where i can found and modify "by hand" these parameters in th E61.
Manny thanks in advance.
(Sorry for my so bad English)
Jean Louis
PS : I have read, here, that other peoples have same problem without any answer from Nokia (Is it Nokia forum here ???)
So if its not the good place for this problem, where to go ?
Message Edited by silver84000 on 22-Nov-2006
02:32 PM
Message Edited by silver84000 on 22-Nov-2006
02:33 PM

This is a user to user assistance forum, the Nokia staff only act as moderators rather than give advice or fixes.
I have full End User experience of:
5510, 3210, 3310, 3510, 3510i, N80, N80IE, 7610, 2610, 1208 and current E90.

Need Open Source Sip client(user agent)

Hi,
Can any one tell me where to find and open sourse sip client which has the facility to call PC to PC ,PC along with IM(messanging).
I have tried the SIP-Communicator that is not good.
If You have the code please forword me or let me know .
My mail id is [email protected]
pramod.

hello,
for the last 4 weeks I struggling to know how to make my mjUA working for voice communication. Thanks God, I found your message that sent last year.
Now I use a free VoIP registrar server (just like free world dialup), but my mjUA can't make any call through that server although my mjUA can register to that server successfully.
my question is:
- is mjUA can be run using free registrar server? or I must use the MJSIP server?
- for the audio tools, I use the JMF (not RAT) without any change on mjUA. Is that right choice?
- can you send me the really working configuration file (such as a.cfg on config folder)?
I'm so desperate because this UA's is the key for my final project. And I hope you can help me.
Best regards,
paulus ivan

E61 SIP client

Similar Messages

Maybe you are looking for