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:\>
Similar Messages
-
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.
-
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 appreaciateI 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 -
Voice gateway, SIP Options ping without SDP
Hi all.
I see following SIP options call-flow for sip options ping:
17907581: Jan 16 08:11:31.057: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
OPTIONS sip:10.250.20.25:5060 SIP/2.0
Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77
Call-ID: [email protected]
From: <sip:[email protected]>;tag=sbc0800uf7upc43
To: <sip:10.250.20.25>
CSeq: 1 OPTIONS
Max-Forwards: 70
Content-Length: 0
17907583: Jan 16 08:11:31.057: //28111388/1CCF073F9ED3/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.11.3.17:9256;branch=z9hG4bKabuf3p5e3es57i595u9viaf77
From: <sip:[email protected]>;tag=sbc0800uf7upc43
To: <sip:10.250.20.25>;tag=BC08FBDC-1F16
Date: Fri, 16 Jan 2015 08:11:31 GMT
Call-ID: [email protected]
Server: Cisco-SIPGateway/IOS-15.4.20141104.060737.
CSeq: 1 OPTIONS
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Accept: application/sdp
Supported: timer,resource-priority,replaces,sdp-anat
Content-Type: application/sdp
Content-Length: 363
v=0
o=CiscoSystemsSIP-GW-UserAgent 1616 9170 IN IP4 10.13.4.43
s=SIP Call
c=IN IP4 10.13.4.43
t=0 0
m=audio 0 RTP/AVP 18 0 8 9 4 2 15
c=IN IP4 10.13.4.43
m=image 0 udptl t38
c=IN IP4 10.13.4.43
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:320
a=T38FaxUdpEC:t38UDPRedundancy
I know that this is normal behavior for voice gateway(3925 in our case).
Is it possible to disable SDP in 200 OK message going out to ITSP. They said that their PBX can't recognize 200 ok with SDP as reply to sip Options message.Oleh,
I am not sure you can do this..
RFC3261 states the following:
The SIP method OPTIONS allows a UA to query another UA or a proxy
server as to its capabilities. This allows a client to discover
information about the supported methods, content types, extensions,
codecs, etc. without "ringing" the other party. For example, before
a client inserts a Require header field into an INVITE listing an
option that it is not certain the destination UAS supports, the
client can query the destination UAS with an OPTIONS to see if this
option is returned in a Supported header field. All UAs MUST support
the OPTIONS method.
If the response to an OPTIONS is generated by a proxy server, the
proxy returns a 200 (OK), listing the capabilities of the server.
The response does not contain a message body.
So it looks like your ITSP hasn't designed their system based on this RFC. When you send an OPTIONs message, the response has to include the capabilities the other party can support.. -
Best way to implement SIP Options Pings on a SIP Trunk
I wanted to see if anyone had suggestions on the best way to configure SIP Options Pings.
Typically I would configure them per dial-peer. However, I really want to do it per destination IP address. I do not want a SIP Options ping for every single dial-peer being sent out every X seconds.
Example:
In my case I have 4 SIP trunks in the same CUBE. Each pointing to a different destination IP. There are 6 dial-peers per SIP trunk. I really do not want 24 option pings going out every X seconds. I guess I've never actually did a debug to see how many pings are going out at a time but I am assuming it sends one for each dial-peer or does it?
If I am correct in my assumption, is there way to only send one ping per destination IP and if that single IP goes unresponsive then all 6 dial-peers go down?Hello,
The OOD option ping is sent per dial-peer to the destination.
Restrictions
•The Cisco Unified Border Element OOD Options ping feature can only be configured at the VoIP Dial-peer level.
•All dial peers start in an active (not busied out) state on a router boot or reboot.
•If a dial-peer has both an outbound proxy and a session target configured, the OOD options ping is sent to the outbound proxy address first.
•Though multiple dial-peers may point to the same SIP server IP address, an independent OOD options ping is sent for each dial-peer.
•If a SIP server is configured as a DNS hostname, OOD Options pings are sent to all the returned addresses until a response is received.
•Configuration for Cisco Unified Border Element OOD and TDM Gateway OOD are different, but can co-exist.
//Suresh
Please rate all the useful posts. -
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 -
Cisco CME to Cisco UBE Options ping no-worky....
Did anyone get SIP options pings working on a pure Cisco router (UBE to UBE and UBE to CME) environment? I have a pure Cisco 15.1 network that we are playing with option pings on. I see pings go out and 200 OK responses, but dial-peers are still busying out. Have tried both default and example settings from Cisco UBE deployment guides. Routers are working with options pings from external ITSPs, so know options replies are working and are supported. So, what's the magic sauce on Cisco IOS 15.1 for SIP options-keepalive?
dial-peer voice 40221 voip
corlist outgoing peer-internal
description \\\ 4385 - 4389 \\\
translation-profile outgoing ToCUCME_NANP_PrefixedE164
preference 1
destination-pattern 438[5-9]
session protocol sipv2
session target ipv4:10.5.5.1
voice-class sip early-offer forced
voice-class sip options-keepalive up-interval 12 down-interval 65 retry 3
dtmf-relay rtp-nte
fax-relay ecm disable
no fax-relay sg3-to-g3
fax rate 14400
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-throug
no vadThere are no firewalls or NAT between and when using "debug ccsip message" we see option sent and received and replies on both sides.
-
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
JuanDid you get a SIP client for the N91? if so where from.
-
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?? -
Why we use different clients in production?
Why we use different clients in production?
Regards,
SubhasishHi,
You usually will have only one client in production.
Its in either QA / DEV environments you will have multiple clients.
DEV - You have one for Config and once for custom development
Regards,
Ravi
Note :Please mark the helpful answers -
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
AlainThere 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 -
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).
TomDoes 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.
-
HT5781 why does my iPhone 4 ping constantly and not connect to my macbook air
why does my iPhone 4 ping constantly and not connect to my macbook air in itunes
Hi, pinkshaz.
You may find this article helpful when troubleshooting iTunes not recognizing your iPhone. If you have any security software, disable this software and test the results.
iOS: Device not recognized in iTunes for Mac OS X
http://support.apple.com/kb/TS1591
Cheers,
Jason H. -
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.
Sacolets 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. -
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
Maybe you are looking for
-
Customized DATA DEFINATION,TEMPLATE FOR R12 WIRE PAYMENT
I have requirement to use lot of custom value in my R12 wire payment report. I have done the following 1. i have creaed one procedure and registered it as concurrent program 2. created data definatin and map it wiht Concurrent program name 3. created
-
Hi, I want to execute a query on Runtime repository user. I need to know: Design Repository User (own of mapping execution error) , Mapping Name and Error Message. With this query I have the Mapping Name and Error Message. BUT HOW CAN I OBTAIN THE De
-
Positive & Negative Time Management
Dear Seniors, What is positive & negative time management? Which time management status do we use for these? What r time pairs & where can these be configured? Thx & Regards WARNED-> No more basic questions please.
-
When I hit mail to check my e-mail it keeps telling me the password is incorrect and its not!
When I go to check my email a box pops up telling me the password is wrong and it's not! I have tried shutting my phone completely off and turn it back on and it still does it!
-
Need help with nonfunctioning SB Live! 24 bit in Vista 32 b
I'm trying to get line-in working on my SB Li've! 24-bit internal card with Vista 32-bit Home Premium. In Control Panel> Audio Devices>Sound the following devices show under the recording tab: Microphone, Line-In, and S/PDIF_In. The only one that see