BT Blocking VOIP Servers? (Mumble)

I'm hosting a mumble on my Raspberry Pi Micro Computer and after setting it up and testing it out, after I quit mumble next time I try to reconnect it says "Server connection failed: Connection refused." which translates to 'something is blocking this'.
It's not a firewall on my local machine(s) as i've tried it with and without it being enabled and it's not the ports not being opened, I have opened all the ports for both TCP and UDP.
This brings me to the conclusion that the problem lies with BT Blocking the port 64738 (Mumble port) or voip servers all together.
Can anyone else shed light on this issue?

BT Parental controls can cause issues.
If you do not need them, then you need to uninstall them.
There are some useful help pages here, for BT Broadband customers only, on my personal website.
BT Broadband customers - help with broadband, WiFi, networking, e-mail and phones.

Similar Messages

Blocking Ad servers via etc/host file not working

In Snow Leopard I add a list of ad servers and direct them to 127.0.0.1 via the /ect/host file.
I've configured the host file on my mac mini server and it doesn't work.
i just want to block ads when using browsers ON the Mini.
How can I achieve this?

Yeah; that's an old-time Windows and Unix hack.
There are ad-blocking extensions for all major web browsers, and those are easier to manage and maintain.
Off the top, I'd have to check the search order to see if I was getting a response from remote DNS or from /etc/hosts here.
If you're running a Mac OS X Server box, you're hopefully also running DNS services, and if you're running DNS services, then that's going to be where this hackery gets implemented. Add bogus zones. And if DNS gets an answer from its cache, I wouldn't tend to expect it to go read /etc/hosts (for performance reasons), for that matter.
There may well be an ad-blocking DNS server around you can forward queries to, too; I haven't looked for that.
If you make changes to your local /etc/hosts setup, try issuing the following to clear the cache:
dnscacheutil -flushcache
I'd recommend installing an ad-blocker. /etc/hosts is an old-school approach and can end up causing weirdnesses on errors, particularly if your box is (also) a DNS server for your LAN. (Also realize that Apple isn't documenting the /etc/hosts file in the Mac OS X Server 10.6 networking manual.)

My phone is blocked from servers and im tryn to switch i to straight talk?

i had to do a system reboot before i could even view apps. so a few days ago i bought a straight talk activation kit and it worked for a few hours. but now it wont take service and when i try to call it say phone has been blocked from service network.

Excuse me?
A: Where did you get the phone?
B: What is the EXACT error message you get?

Airport blocking VoIP soft phone

When I bypass the airport and patch straight via ethernet cable into my modem, my softphone works fine. When trying to use it via the airport, no luck.
The modem manufacturer - ZyXel - said the airport is acting as a second router, and I've got to forward the airport to 192.168.2.2 I tried a few things in the airport set up utility, and all that did was shut me down completely.
It's very frustrating because I can go just about any place and use this softphone, but not at home via my airport network.
Can anyone explain to me how to do this? This is territory well out of my range of expertise.
Thanks in advance for any help!

Since you ZyXel is acting as a router, you don't need the AirPort Extreme base station (AEBS) to act as one. Configure the AEBS so that it acts as a bridge (disable the option to distribute IP addresses).
Then you won't need to fool with port forwarding... all the ports are forwarded automatically.

N8; How to set up SIP and make (WiFI & 3G)VOIP cal...

N8 only comes with SIP software and you need to download the the VoIP setting software at this link (use VoIP 3.1):
http://www.forum.nokia.com/info/sw.nokia.com/id/d476061e-90ca-42e9-b3ea-1a852f3808ec/SIP_VoIP_Settin...
You can use the N8 web browser and load directly from the site to the phone.
then go to:
Menu>settings>connectivity>Admin.Settings>SIP settings
option>New SIP Profile>use default profile.
This window needs to be configured with your VoIP provider setting which may be found on their site or just do a search. I used the settings from my previous phone. I have Gizmo VoIP and will be using the setting for this service as an example to set up Wifi and 3G SIP setting
Profile Name> Gizmo, (assign the name of your provider e.g.. Gizmo, Google talk, etc)
Service profile> IETF
Default Destination> WiFi
Public username: sip:[email protected] (Input your VoIP provider info)
Use compression> No
Registration> When needed
Use security> No
Then goto Proxy server:
Proxy server address> siproxy01.sipphone.com (use your provider proxy address setting)
Realm> proxy01.sipphone.com (use your provider setting)
Username> (use your username)
Password> (enter your password0
Allow loose routing> Yes
Transport type> UDP
Port> 5060
Go back and then go to Registrar server:
Registrar server address> siproxy01.sipphone.com (use your provider Registrar setting)
Real> proxy01.sipphone.com (use your provider setting)
Username> (use your username)
Password> (enter password)
Transport type> UDP
Port> 5060
Again folks above is just an example. You must use your provider settings.
Then when you are done go back to:
Admin setting> Net setting>Advance VoIP setting>Create new service> choose the SIP profile shown, e.g. Gizmo
This will configure the VoIP setting with the particular SIP profile. It will also add a SIP Tel profile Pane in your Contacts folder, Internet call option to the contacts list and Phone log.
Now go to the SIP pane in Contacts folder, sign in and then make a test call . To make a SIP call just go to a contact and choose Internet call. If you configured everything correctly, it should work and you are done.
For 3G connection, follow above direction and create new SIP profile with new name (e.g. Gizmo (3G). All settings are same except:
Default destination>Internet
Then set up the VoIP setting with the new SIP profile by going to:
Admin setting>Net setting>Advance VoIP settings>Create new service> Choose the SIP profile you just configured e.g. Gizmo (3G)
Then:
Admin setting>Net setting>Advance VoIP settings>VoIP service>choose the 3G SIP profile, e.g. Gizmo (3G)>Profile setting>>>>>
Go down to AWCDMA> set it to "on"
Exit out of everything. Go to SIP (3G) pane in the Contacts folder. Sign in and make test call.
I have configured many Mobile phones and found the N8 somewhat touchy. But, the sound quality on Gizmo VoIP using both WiFi and 3G was great, no echo or any problems.
Useful hints:
If it does not connect or make a call, you need to check the SIP setting for any error. If you find any problem there, you must first remove the SIP profile in the Net setting. Then go to the SIP setting and correct the problem. Double check everyting, go back to Net setting>Advance VoIP setting>Create new service and add the SIP profile. The reason is that when you Configure a VoIP setting with a SIP profile, the SIP info is permanently loaded and will not change even if you make any change in the SIP profile. So any time you change anything in the SIP profile (even the password) you always need to remove, make correction and then reload the corrected SIP profile to the VoIP setting.
There are also few Mobile service providers who have blocked VoIP on their data network service. So, just check with your service provider. In US both AT&T and TMobile have open network for VoIP.

Hi
I have an unbranded N8 and am trying to get VoipCheap to work.
I have tried to set up VOIP but still facing a problem or 2 so please assist me.
I have an active account via VoipCheap.com with a profile name and password and it is verified.
1/ When I look in Admin Settings / Sip Settings it shows VoipCheap with "not registered" under it..why?
My settings when I click on VoipCheap are:
profile name: VoipCheap
Service profile: IETF
Default destination: Internet
Public username: sip:[email protected]
Use compression: No
Registration: When needed
Use security: No
Proxy server address: sip:sip.VoipCheap.com
Realm: VoipCheap.com
Username: [email protected]
Password: ****** (I have entered my password)
Allow loose routing: Yes
Transport type: UDP
Port: 5060
Registrar server settings same as Proxy server settings.
I have also gone to Net settings / VoipCheap settings and see:
Username: sip:[email protected]
Password: ***** (I have entered my password)
Default service: No
Service connectivity: Internet
In Contacts I have a VoipCheap tab with the icon for internet calling and it says
Sign in (example)
but when I click on this it shows connecting (via my internet connection) but hangs there - it does not seem to connect............
what do you think I am doing wrong, please ????????????????

VoIP chalenging questions

Dear All,
I've got some chalenging questions about implementing Cisco VoIP and I looking for some fair answers:
1. What visible indications are invoked if IP phones default to survivable telephony mode?
2.How long will active calls last if primary call control is lost?
3.How does VoIP solution provide effortless IP phone addressing without having to change the addressing scheme of the existing IP data network?
4.Under what circumstances would a telephone outage affect the LAN/PC communications to the network?
5.Does the system support clustering? If so, across how many systems?
6.What is the highest number of IP devices currently supported under a single instance of the software? How many IP devices are supported by a cluster?
7.Does the system have a web-based or GUI-based administration tool? Is this included in the base price?
8.Can a single administration portal/tool be used to manage all systems within an enterprise simultaneously? If not, explain the administrative process?
9.How does the system provide for fault tolerance? f.i. Is it ?hot standby redundancy?
10.If/when a switchover event occurs, describe the events and detail what a subscriber and caller would experience.
11.How are databases synchronized?
12.How is time clocking designed?
13.How is clocking synchronized?
14.What virus and intrusion protection is included ?
15.What is the response process and what activities are conducted in the immediate identification of new viruses, denial of service and similar attacks?
16.What activities or applications do you recommended / conduct to detect and block VoIP spam?
17.What toll fraud protection is available? Identify its process. How is it managed & monitored?
18.How many pairs of wires are needed to support non- IP telephones: digital telephones, analog stations, terminals and consoles?
19.Identify and describe the distance limitations and wire gauge limitations to your station equipment, consoles, administrative terminals, etc.
20.Do digital or IP telephone ?sidecar expansion? units require local power? If so, identify & explain the requirement.
21.Describe the power requirements for all system components.
22.Wwhat are the two-hour UPS or ?battery? backup capabilities ?
23.What system management diagnostic tools can be use?
a.What types of event logs does the system support?
b.What diagnostic reports are available to aid in isolating faults? Can diagnostics be remotely accessed?
c.Are the system?s diagnostic tools SNMP compliant?
24.What specific VoIP system diagnostic & management tools can be use?
a.What types of event logs does the system support?
b.What VoIP reports are available to aid in isolating faults? Can diagnostics be remotely accessed?
c.Are the system?s diagnostic tools SNMP compliant?
25.What remote diagnostics are available? Can administrators see and access any alarms or alerts on the system from remote terminals?
26.What system monitoring report(s) are available ? How about any third party monitoring products for the followings:
-Status trunking
-Real-time traffic
-Status of call center
-Status of call center components
-Status of all routing components
-Status of all remote components
-Status of individual stations
-Status of all gateway ports
-Call trace capability
I would like to thanks in advance to each of you for your time and consideration.
Regards,
Mike

Someone would spend a day typing everything up for you. If I were you, I would pick up a CallManager book. "Cisco IP Telphony: Planing, Design, Implentation, Operation and Optimization" There is also "Cisco CallManager Best Practice."
Or call your local Cisco Sales guy, they have local SE's that can give a briefing of the product in more details.
Most of know all the answers to those questions, but it would take any of us 2-3 hours to pull up. The books Cisco have out are great for reference and quick lookups for this stuff.
good luck!

Connecting n95 with Sipgate VoIP to Talk Talk broa...

I'm helping a friend set up her n95 on her talk talk broadband. the handset is successfully set up with sipgate and it works at my house, I've opened the ports on her Thompson Speedtouch 780 router as suggested by sipgate but still nothing. her TAlk Talk router was rubbish and didn't show up in wi-fi searches half the time so i got her the Thompson one. either way, i couldn't get sipgate to connect with either router. yet the n95 connects to the net without a problem? does talk talk block VoIP from its broadband?
Nexu 5

you can try the discussion forum in www.forum.nokia.com also

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

Linkwall not blocking URLS by IP address

Hi
We have noticed that linkwall does not block sites if you use the IP address rather than the DNS name. For example in the linkwall.lst file Poker.com - The home of online poker games, rules & tournaments will block that site but not the IP address of the site which can be obtained by dns lookup.
Rules placed in the BM acl list in nwadmin do block the corresponding IP address.
Does anybody know if this is working as designed - it doesn't really provide much protection this way.
Many thanks
Kerry

HI Gonzalo
Thanks - I have already tried the proxy.cfg with the DonotSendIPToACL set to 1 and 0 (restarted with -cc etc) and it made no difference. My understanding was that the switch only affected sites that were multihomed.
Never mind - it certainly doesn't do the lookup with the third party tool so we looking into other options.
Thanks for the input though
Kerry
Originally Posted by mysterious
KPR wrote:
> Hi
>
> We have noticed that linkwall does not block sites if you use the IP
> address rather than the DNS name. For example in the linkwall.lst file
> 'Poker.com - The home of online poker games, rules & tournaments'
> (Poker.com - The home of online poker games, rules & tournaments) will block that site but not the IP address of
> the site which can be obtained by dns lookup.
>
> Rules placed in the BM acl list in nwadmin do block the corresponding
> IP address.
Probably you're using an empty proxy.cfg file.
> Does anybody know if this is working as designed - it doesn't really
> provide much protection this way.
About linkwall, Marcus already answered you.
About bm, this behavior was changed to stop blocking multihoming
servers. Multihoming servers are servers that with one ip address, they
host various web servers. For example, a server with ip address 10.1.1.1
is hosting Poker.com - The home of online poker games, rules & tournaments, www.penguins,com and www.dolphins.com
So you block Poker.com - The home of online poker games, rules & tournaments, bm resolves this domain to 10.1.1.1 so this
ip address is blocked as well. Now you try to access penguins.com
and you're block. but you did not block this site. So to avoid that, a
new switch was introduced for bm and the switch is placed on the
proxy.cfg file:
DonotSendIPToACL=1
with this switch, only domain name will be blocked. Without it or set to
0, both, domain name and ip will be blocked, so you can run into the
issue with multihoming servers.
Gonzalo
> Many thanks
>
> Kerry
>
>

VoIP connection to company server

Hi there,
When reading the specs of the IPhone 3G I was interrested in its capability to connect to VoIP servers. I noticed there are lots of apps around which let it connect to skype like services. In other words: services for which you need an account and you place calls via the Internet.
This is not what I am looking for.
What I need is a replacement for the Nokia E series.
With the Nokia E series I can make a VoIP profile connecting to our company VoIP server. The profile becomes an intergral part in the phones methodes of placing and receiving calls.
This way I can place and receive calls through our company VoIP server when in range of our Wifi network. When out of range it switches to the network of the telecom provider (like Vodafone).
Is this posible with Apple IPhone?
Peter

Seems it is called ISip nowadays.... looks good.
Now all I need is an IPhone to actually test it .... thanks

What are the iMessage servers?

I use SelfControl to block my email and websites that distract me. I want to block the servers used by iMessage as well. How can I find out what server my iMessages go to?

HI,
The server is listed in the com.apple.imessages.bag.plist
This is in ~/Library/Preferences
10:15 pm Thursday; October 3, 2013
 iMac 2.5Ghz 5i 2011 (Mountain Lion 10.8.4)
 G4/1GhzDual MDD (Leopard 10.5.8)
 MacBookPro 2Gb (Snow Leopard 10.6.8)
 Mac OS X (10.6.8),
 Couple of iPhones and an iPad
"Limit the Logs to the Bits above Binary Images."  No, Seriously

Block Skype Ads Quick and Easy

Hi everyone, I was trying to find a way to block all these annoying ads that the new Skype has introdused.  There where a couple of ways to actually achieve this but most of them where not working after Skype was updating. Below is the way that works regardless the Skype version as it 'hits' directly in the source, the ads themselfes.  1 Press Windows+R and type C:\Windows\System32\drivers\etc\hosts   OR 1. Navigate to the folder C:\Windows\System32\drivers\etc\ , right click on the hosts file and hit Open (or edit with Notepad++ if you have it)2. Select Notepad or any other text editor you prefer
2. Add at the bottom of the file the lines below: # Skype Ads Blocker
127.0.0.1 rad.msn.com
127.0.0.1 live.rads.msn.com
127.0.0.1 ads1.msn.com
127.0.0.1 static.2mdn.net
127.0.0.1 g.msn.com
127.0.0.1 a.ads2.msads.net
127.0.0.1 b.ads2.msads.net
127.0.0.1 ac3.msn.com# End Skype Ads Blocker
3. Save and restart your Skype app.
What you are basically doing is blocking the servers that Skype is pulling the ads from.  If you have any problems after applying this fix or you want to see these beatifull ads again (you must be drunk to do so! ) simply remove the lines you've added in the host file. Hope this helps someone. PS1: Tested to version 6.16+PS2: This should not breach the Terms and Conditions/User Agreement with Skype as you are NOT manipulating or editing Skype itself. You are simply blocking traffic from some specific servers ( just mentioning in case we have people starting a catastrophe reply marathon )

etc seems to be hidden within the drivers folder. In Windows Explorer you can click on the address bar and enter the path to that folder manually:C:\Windows\System32\drivers\etc Note that you might need to edit the host file using admin rights. And btw: If you don't mind losing your "Skype Home" than one line is sufficient:127.0.0.1    apps.skype.com

Which VoIP provider is airport compatible with

I Need to replace my router and want to go with the AirPort Extreme. I have a VoIP bridge which I confirmed with 8x8 will not work. Are there any other VoIP providers whose bridges are compatible with the AirPort Extreme?

The problem is apple do not make their routers voip aware.. they also do not make routers with standard upnp to open ports.
I think you can get any voip setup to work.. but it is up to you to manually forward the ports. Apple will give you no help. But nor does it block voip packets.. if that is any consolation.
If it is major to you to run voip.. do not use apple routers.. you can buy an extreme and use it bridged.. as long as it doesn't route it cannot block anything.
Apple is a closed garden.. they build stuff so their own products can work.. and the rest of the world be blowed.

Apple blocked Face Time in Middle East

Hello there iPhoners
i heard that the face time will not work in some countries such as saudi arabia and united arab emirates
i want to know if my iphone which i bought from the UK will be blocked from using face time or not because i have bought it for using this feature

The phone does not block FaceTime. It should not matter where you purchased your iPhone. Internet providers and carriers block the common ports that VoIP uses. In reality Apple isn't blocking FaceTime some countries don't allow or regulate VoIP. It is unfair to only bring attention to the countries of the Middle East. There are other countries that also block VoIP.
I am not sure if Saudi Arabia blocks VoIP. I have read conflicting articles on the subject.
http://www.brighthub.com/office/collaboration/articles/61546.aspx

Conflict with ISA570W Remote VPN

Ok... this one makes absolutely no sense. I'm hoping someone can help.
My customer's IP phones stopped working when I enabled Remote IPSec on the ISA570W.
Here's the configuration:
When the Mitel IP phones power on, they acquire an IP address from the hosted VoIP server via DHCP (option 128) then download the configuration file via TFTP. When I enable Remote IPSec on the ISA570W, the phones obtain their address, but stall during the TFTP transfer. If I disable Remote IPSec and power-cycle the ISA570W, the phones start up normally. If Remote IPSec is enabled and I disconnect the WAN1 interface, the phones start normally.
I fail to see how enabling Remote IPSec on the ISA570W causes a communication failure between the IP phones and the hosted VoIP servers.
The phones are not communicating through the ISA570W. The path is: IP Phone<-->LAN<-->Cisco 2430<-->Host
Any ideas?
Thanks,
Rick

Nagaraja,
The VoIP solution is hosted by a third party (Broadview Networks), so I have limited information on the configuration of their Cisco 2430.
Here's what I can provide:
LAN
192.168.0.0/24
ISA570W
LAN
    VLAN 1 (Default - GE2-GE9)
      IP Address: 192.168.0.1/24
WAN1 (GE1 - Internet - Fairpoint FTTP)
    IP Address: 71.***.***.39/24
    Add'l IP: 71.***.***.139/24
WAN2 (GE10 - Internet - Broadview Cisco 2430 Data interface)
    IP Address: 64.***.***.138/31
Note: WAN1 is used primarily for Internet access and inbound connections.
           WAN2 is used for site-to-site IPSec over Broadview Internet to remote office.
Site-to-site IPSec: 192.168.0.0/24 <--> 192.168.16.0/24
Cisco 2430
WAN: T1 to Broadview Networks
Data interface (fe?/?)
    IP Address: 64.---.---.137/31
Voice interface (fe?/?)
    IP Address: ???
TFTP Server: 10.161.0.32/??
Phones are assigned IP addresses on network 10.160.90.0/?? via DHCP (Option 128)
Servers, computers and phones are connected to the same physical switches. This is necessary because we use the passthrough port on the phone for the computers.
When Remote IPSec is enabled on the ISA570W, phones successfully acquire DHCP and begin the TFTP transfer, but stall after a few blocks. In rare cases, a phone will complete the TFTP transfer but hang during initialization.
Here's a sample capture:
10.160.90.43 10.161.0.32 TFTP: Read Request - File: MinetIp5220Dpl.bin, Transfer Mode: octet blksize: 4096 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Option Acknowledgement - blksize: 1024 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 0 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 1 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 1 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 2 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 2 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 3 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 3 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 4 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 4 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 5 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 5 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 6 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 6 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 6 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 6 {UDP:3132, IPv4:3129}
10.161.0.32 10.160.90.43 TFTP: Data - Block Number: 6 {UDP:3132, IPv4:3129}
10.160.90.43 10.161.0.32 TFTP: Acknowledgement - Block Number: 6 {UDP:3132, IPv4:3129}
Note that in this example, block six continues to retry indefinately.
Any one of the following actions will clear the problem:
Disconnect the ISA570W from the LAN
Disconnect the ISA570W WAN1 interface
Disable Remote IPSec and power-cycle the ISA570W
Here's the Remote IPSec configuration:
Group name: M*******s_VPN
WAN Interface: WAN1
IKE auth method: Pre-shared key
Password: ************
Mode: Client
Pool range for client LAN: 192.168.0.176 - 192.168.0.191
Client internet access: Create NAT rule... enabled
WAN failover: Off
Zone Access Control: Only LAN and WAN permitted
Mode configuration:
    Primary DNS: 192.168.0.19
    Default Domain: m*********c.com
    Split tunnel: On
      Protected network: 192.168.0.0 255.255.255.0
Based on the capture, the path between the TFTP server and the phone is unaffected. The TFTP traffic is UDP. Is it possible it's getting corrupted?
Thanks,
Rick

BT Blocking VOIP Servers? (Mumble)

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