WRT110 voip disconnect
Very disappointed with Linksys routers. My WRT110 router disconnects my VoIP calls after call is answered.
Updated to Firmware Version: 1.0.04 (Got it somewhere on the net. Linksys does not even support it yet!) but the problem resists.
Tried port forwarding and port triggering (as other Linksys routers have this issue) but no luck.
Anyone knows why this happens?
Solved!
Go to Solution.
Thank you for your reply. I have tried it with and without STUN server, forwarded SIP/RTP ports to the ATA (SPA2102), tried ATA in DMZ with no luck. I guess this is happening with SPA2102 because my Siemens C410IP works without any problem. And I did not have this problem on my WRT-160N with DD-WRT firmware.
Linksys must do something about WRT router production line. WRT-160N caused me a lot of problems before using dd-wrt firmware on it.
Similar Messages
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I purchased a WRT110 router about a 18 months ago. I am currently experiencing an issue where the wireless connection to the router will become randomly erratic. My internet connection will slow down or drop entirely. Sometimes my laptop will disconnect from the router entirely. I don't think it's the wireless card in my laptop since I have seen the issue on my desktop using a wireless N USB adapter, my Nintendo Wii, and my wife's Macbook, all within the last week.
I am positive it is the router. I set up a coninuous ping command to the router accross my laptop's wireless using "ping -t 192.168.2.1" in a command prompt. Most of the time, the round trip is 1 to 3 ms or <1 ms. When I am experiencing the connection issues, the ping time will rise significantly, sometimes going as high as 2000 - 3000 ms. About 25% of time, the ping will just time out. When this starts happening, the only way I have been able to restore proper connectivity is to manually reboot the router. The issue eventually re-appears. Sometimes in a few days, sometimes in as little as five minutes. I've gone so far as to restore the router to factory defaults and reconfigure it, but that has not made a difference.
I also experienced this exact issue about two months after I bought the router. I was trying to find the receipt so that I could return the it when the issue suddenly stopped and the router worked perfectly for better than a year, until now.
Any ideas?yup... I agree. you might have an outdated firmware.
Here's the link to the download the latest firmware:
http://homesupport.cisco.com/en-apac/support/routers/WRT110/download
"Don't fix it if it ain't broken." -
Linksys WRT110 keeps disconnecting
I have a Linksys WRT110 and it worked fine for about the first year or so, but recently it has been intermittenly droping the signal to the point of no connection at all. Is there an update or anything I can do about this? I contacted tech support, and conveniently since my warranty is up I have to pay for their help with this...
(Mod note: Off topic remarks removed. Thanks!)It would drop it when I was using my iPhone and when my wife used her laptop. It was very intermittent and never stayed offline for the same amount of time. Sometimes it would be less than a minute, sometimes more. Occasionally I had to turn wifi off on my phone, then turn it back on just to get the wifi going again.
It was dropping the wired connection a few months back, but it got solved on here. Had to to a complete reset. -
While using ANY of the three computers on my network, 1 wired XP SP2 box and 2 wireless Vista laptops (1 Vista Business, 1 Vista Home Premium), my WRT110 just completely drops out for about 2 minutes and then comes back like nothing was ever wrong. I notice this especially on both wireless laptops, it appears that for a brief moment the entire connection drops, then the local connection returns but the internet drops for about 2 minutes. This is a brand new WRT110 and I'm connected on a brand new HughesNet satellite modem (Modem to WRT110, WRT110 to my computers). Any suggestions? This is really frustrating. Thanks!!
I have having a similar issue.
I have a brand new WRT110 and a brand new Motorola SB5101 Surfboard cable modem.
When I bought the WRT110, I got online with live chat, and a very helpful gentleman helped me get everything set up ... that was two or three days ago.
Yesterday, I found out from my cable provider that my cable modem was very outdated, so I upgraded to the SB5101. I checked all the settings in my router, per the chat log session that I got from the previous day, and everything was great.
But my connection drops repeatedly throughout the day ...
I have a Vista Home wired into the router, and a Vista Home laptop that connects wirelessly. The laptop is less than 30-40 feet from the router, so I know the issue is NOT with the distance.
I do also have my DirecTV DVR wired (via a cable) into the router, and it, too, has trouble connecting and accessing data from my computers ...
I have the latest firmware on my router.
It is frustrating, especially on the laptop, because when I restart the laptop, I often get limited connectivity or no connectivity.
Any suggestions? -
WRT110 Bricked after fw update to 1.0.07
My wrt110 started disconnecting twice and even three times a day for the past week or so. After getting tired of it I decided to look for a fw update. Hold and behold there was, from 1.0.05 I was going to upgrade to 1.0.07. Well halfway through the process using the web-console connected via ethernet & ie8 the freaking upgrade stopped at 80% and the browser page went blank. Immediately I knew I was probably out of luck, looked at the wrt110 to see a blinking power light. I've done all the resets in the world inluding the so called 30-30-30 to no avail.
(Mod note: Edited for guideline compliance.)
Message Edited by iamgiantherockstar on 03-30-2010 09:08 AMAs the power light on the router is blinking,the router seems to be bricked.Try this link to Unbrick the router and try to upgrade the firmware again on the router.
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Do I need a service call to disconnect the ONT from my house wiring after going from FIOS to VoIP?
I recently converted from FIOS to cable phone service (VoIP) (my FIOS contract was up and the cable compan offered me a better deal over the next two years -- no complaints about FIOS itself). Since I converted, I've been having all kinds of problems with the phone service, and all of the problems exist only when I use the house wiring -- when I connect a phone right to the cable modem, it works perfectly. I disconnected the ONT's battery backup unit and unplugged it, but that caused me to lose the dial tone after a few seconds -- I got nothing, and then a fast busy (again, this happened only when I was connected to the house wiring -- it didn't happen when I used a phone that was connected directly to the cable modem). Putting the battery back into the ONT's battery backup and plugging it back in corrected that issue. Additionally, I don't get a triple beep when I pick up the phone if there's a voice mail, nor does the phone's display screen indicate that there is a message waiting. In short, it appears that the house wiring is still connected to Verizon's system. The cable company says the house wiring isn't connected to the old (analog) Verizon box outside the house, but it says it can't touch the ONT. Do I have to have Verizon come over to disconnect the house wiring from the ONT, or can I do it myself? If Verizon comes to do it, will I get charged? Thanks. David Ross {edited for privacy}
The ONT should have a place for customer connections. Either a RJ-11 plug that can be unpluged or wires on red and green terminals that can be disconnected and isolated with tape or wire nuts. Sorry to see you go, I have been happy and will never go back. Well never is a long time.
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Is there a way to (temporarily) disconnect from VoIP (for the purpose of reconnecting) without having to change the login type from automatic to manual? If I disconnect VoIP or WLAN, the phone always wants to change things to manual as well. I don't want that.
Solved!
Go to Solution.You should be able to force VoIP to re-register by manually disconnecting the active WLAN connection that VoIP is currently using.
Go to Connection Manager application (usually found under Menu -> Tools or Menu -> Connectivity) and select Active Data Connections -> choose the WLAN connection which VoIP is using -> select Options -> Disconnect.
At least in my case forcing WLAN connection to drop temporarily like described above does not change VoIP settings (from Automatic to Manual) and as long as I have SIP/VoIP originally set to "Automatic/Always on", have periodic WLAN scanning enabled and WLAN network coverage remains available then VoIP client (E71) does automatically open up a new WLAN connection and re-register itself on the VoIP service. -
Just installed a wrt110 over christmas. Used an XP system to set it up and had no issues. Added some HP printers, another XP system, and a new Windows 7 notebook with no problems. Then added a Vista notebook. Now it appears everytime my daughter turns it on and connects to the network the 110 just locks up. I have to do a repair and it finally comes back up. I have ipv6 disabled on all systems that have that as an option. Seems like even though this is a "new" device the firmware on it is .4, not .5. Question is will upgrading the firmware fix this problem or not. If not i'm not going to waste any more time on this router becuase there are better ones out there. My wife bought it on sale but I need reliable router since my wife works from home.
I am also having problems staying connected. My desktop still runs Vista but I upgraded my laptop with Windows 7. I had problems staying connected with the ATT aircard but was able to download a new driver and that took of that problem when I am traveling but now I can't stay connected at home. I wish I had never downloaded Win 7. Any suggestions or direct me to a driver, etc. so that I can solve this problem would be greatly appreciated. Thank you.
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New FIOS customer with dropped VOIP calls and Internet connection
I am a new FIOS customer. Got my 50/25 connection a week ago, switching from a TWC 6/1 connection. Ever since the new connection, I've had numerous issues.
My VOIP (Ooma) connection constantly drops and re-connects during conversations
I've had random Internet connection losses, which picks up again after a few minutes
My home alarm starts chirping every once in a while
I've contacted Verizon several times due to these problems and have received varying answers with no resolution of the problem.
The first time I spoke with support, the tech logged into my router and changed the WiFi channel saying that would fix the problem. It didn’t.
The second time I contacted them, the tech ran a bunch of diagnostics and said everything looked fine so it must be an IP address conflict with my devices, because I had a couple devices using static IP addresses. He said everythinf should be DHCP and the last two digits could not be higher than 99 (192.168.1.99). He said FIOS does not support 3-digit numbers at the end.
So I changed all my devices to DHCP and ran some online VOIP tests. It showed a packet loss of 2-5% and MOS score of 1 (which is bad). I was still getting dropped connections, so I disconnected all devices and connected just one computer to the router and tested again. I was still getting packet loss.
Then I called support a third time, this time the tech said there were no 2-digit IP restrictions and that he was detecting there was no UPS baterry backup for the ONT which was probably causing the problem, so he dispatched a field tech to my house.
Today the field tech came (same guy as before), he took one look at the box and said it was too close to my Electric meter and the RF from the meter was causing interference to the FIOS connection and resulting in dropped connection.
He moved the ONT to another location and said that should fix it.
Well, I'm still seeing packet loss and low MOS score when I run the VOIP test.
I don't know how much of what the techs are saying is true and how much is made up stuff.
Has anyone had similar issues and have thoughts on solutions or likely causes for dropped VOIP calls and connections? Could RF be causing this?
I thought going from a 6/1 Cable connection to a 50/25 FIOS connection would be awesome, but this has turned out to be a nightmare, and I may have to switch back to cable if the problem is not resolved.
I would appreciate any help.
Thanks!Don't know where the packet loss is happening. I ran the VOIP test on myspeed.visualware.com and it shows a packet loss of 2-5% at different times and a MOS score of 1.
The report says MOS should be around 4 for good VOIP calls.
The Verizon tech who came to the house just blamed the electric meter box for RF interference and move the ONT farther away.
My concern is that I'm getting different answers from different techs at Verizon.
Regarding IP addresses. The Router shows a DHCP range from 192.162.1.2 to 192.168.1.254 as available for devices on the network. So, if I need to assign a static IP to a device should I use a number below 99 or above 151?
Thanks! -
Help Needed for Resetting VoIP SPA 2100 Codes
Yesterday afternoon, I lost power to my home for a few minutes; until then my phone worked perfectly (for the week that I had it initially installed). I have the Earthlink TrueVoice program with a ZyXEL P 600 series DSL modem and direct Ethernet cable (NO router) to my ATA adapter (Linksys VoIP model SPA 2100). My computer is an HP Pavilion 8755C with Windows Millenium Edition. I have a standard (wire) phone connection. After the power was restored to my home, I tried to use my ATA adapter (Linksys VoIP model SPA 2100) to make a phone call, but there was a five-second delay between when I spoke and when someone on the other end of the line heard me. I contacted Earthlink through online chat and was told to do the following: Dial **** 877778 # 1 to reset the adapter. I did this and nothing positive happened (though I did hear a recorded English woman’s voice repeat the numbers I dialed); I did, however, lose the dial tone and was unable to connect to the Internet with my DSL. So, to conduct more online chat, I re-routed the Ethernet cable to bypass the ATA adapter; I was successful at restoring my Internet, but, obviously, I had NO phone service. More than 8 hours of online chats to Earthlink technical service resulted in failure each time. I swapped ends of my cables numerous times – no success. I removed the power cords to the ATA adapter and to the DSL Modem MULTPILE times for for up to 5 minutes – no success. The lights on the back of the ATA lit properly, but since the first time I tried the **** 877778 # 1 code, the Status light blinks and the light indicating “Phone 1” stays dark (yes, there is astandard phoneline connected to that port). I tried multiple codes to reset the ATA (re-coding instructions are separated by semi-colons): **** 877778 # 1 ; **** 877778 # 1 # ; **** 732668 # ; *** *73738 # 1 . I unplugged the power cord and Ethernet cable (to the modem) on the ATA adapter for about 1 or 2 minutes after trying to reset the code before reattaching everything (power cord always re-attached last). I tried each of these suggestions MULTIPLE times without success. One Earthlink tech wanted more specific information – the MAC address (which I was told is a 12 digit hexa decimal number that starts with zero). I told him what it was. He asked me to try a “ping” test; I was told to click on Start -> Run, type in cmd and hit Enter. A new window would appear and I should type in ping 192.168.0.1 and hit Enter. I never got to the “new window” because after I typed in “cmd” and hit Enter, I received the message “Windows cannot find cmd.” I was told to dial the code * * * * 110 # and to hear my IP address; I heard nothing. I noticed that the last time I tried to dial **** 877778 # 1, I was interrupted before I typed the final “1” because the recorded voice told me “invalid value.” Around 1 a.m. last night I gave up in disgust and went to sleep, disconnecting everything. I have now reconnected the DSL modem and made the direct connection to my computer (the ATA adapter is still disconnected from power and all connecting cables). I am not an idiot and I have some electronics experience (I have wired several home theater systems), but I have little computer experience (and none with VoIP) and right now I am totally frustrated. The tech support at Earthlink is virtually worthless. Please help (and please keep it in simple terms so that an ignorant individual like myself might learn – I need a nice step-by-step method (including times to wait between restarts, what cables/cords to disconnect and the order to disconnect/reconnect them (if that matters) to reset this device) Thanks from a neophyte who is not afraid to be honest and show his ignorance in this area of knowledge.
hi,
i have a SPA-2100 it's a Sipura, and it now asks for a password. I was putting all my settings in for voip provider and now can not get in my admin? i did try the 73738 and that did not work. I was able to put my providers sip and stun all in. is it i'm locked out from my voip provider? If so why would that be, because this is not their device, it's mine? but how do I get back in my admin or can the voip provider I put in, help?
thanks
Message Edited by jokers32463 on 12-22-2008 06:21 PM -
I can no longer connect with my WRT110. If I plug my line direct from a cable modem to my laptop, all is fine. On my desktop not since Charter reset the modem. With the router and the desktop, I get in IP of 0.
I have re-powered everything a hundred times, used the router interface and ipconfig to release and renew the IP and nothing at all works. I have also disconnected the desktop from the router and ran the router to my laptop. Stll no IP. version is 1.0.04Press and hold the reset button for 30 seconds...Release the reset button...Unplug the power cable from your router, wait for 30 seconds and re-connect the power cable...Now re-configure your router...
As your Internet Service Providor is Cable follow this link (NOTE : Don't use your desktop...connect your laptop)Also make sure when you follow the link wireless network connection should be disabled... -
Setting up a VoIP extension on a local network.
With the help of the experts on this board I have successfully set up a VoIP phone extension on our private network. The questions & answers can be viewed at. http://forums.linksys.com/linksys/board/message?board.id=VoIP_Adapters&thread.id=3197 . For the benefit of anyone attempting a similar project, here is the completed setup.
This installation is in a small motel in Te Anau, on New Zealand’s South island. The manager lives off site, and needs to be able to receive calls at night, and also transfer incoming calls to guest’s extensions through the hotels PBX. This necessitates a direct link to the PBX, rather than simply diverting the phone. One solution would have been to lease a circuit from the local Telco, but in NZ, this is very expensive, so another solution was sought. Fortunately there was an established wireless data link between the hotel and the managers residence, so VoIP seemed the obvious choice.
The equipment used is a Linksys SPA3102 connected to an extension on the PBX, and a Linksys PAP2 at the remote end. The setup would work equally well if connected to a phone line, rather than the PBX.
I’ll start the setup with the SPA3102.
Connect the POTS line to the LINE port, and your switch/router to the INTERNET port. In my setup the Ethernet port is not used. Plug a standard phone into the Phone port. This is useful for testing and setting up. It’s not needed afterwards, unless you want a local phone.
Open your web browser, and type the adaptor IP into the address bar. Go to Admin, and Advanced Settings.
ROUTER SETUP
WAN Setup Tab:
Connection type: Static IP.
Static IP Settings: The Network address on your local network (192.168.x.x)
Subnet mask 255.255.255.0
LAN Setup Tab:
LAN IP address: This is automatically selected to be on a different sub net from the WAN. Unless it conflicts with another address on your system you shouldn’t change it.
Enable DHCP: No
(Save these settings.)
VOICE SETUP
System Tab: No Changes
SIP Tab: No Changes
Provisioning Tab: No Changes
Regional Tab: Mostly this sets the dial tones etc to match your local service. Unless you need them to be the same this shouldn’t need any changes
The Hook Flash Timer Min & Max: should be set to the local values. The Defaults (.1 and .9) are OK for North America. Australia and New Zealand use .07 & .13. If you have trouble sending a hook flash, check these values against the local settings.
DTMF playback level should be greater than zero. (I used 3)
(Save these settings)
Line 1 Tab:
I don’t use Line 1 except for testing. During setup the line should be enabled. After the system is running OK, it can be disabled
Line enabled yes
SIP port 5060
Proxies are not used in this setup.
Register: No
Make call without reg: yes
Answer call without reg: yes
User ID: 10? (you can use any number)
Line 1 Tabupplementary services.
Change Call waiting, 3 way Conf, and 3 way call, to no. (These interfere with sending a hook flash)
Hook Flash Tx method: AVT
(save these settings)
PSTN Line Tab
Line enable: yes
SIP Port 5061 (default)
Proxy: proxies are not used.
Register: no
Make call w/o reg yes
Answer call w/o reg yes
Display name: anything you like (VoIP gateway?)
User ID: leave blank
User password: leave blank
Use auth ID: no
Dial Plan 1: (<:*>S0). Switches to the outside line when * received.
Dial Plan 2: (<:[email protected]:5060>S0). 11 is the user ID on the PAP2
VoIP to PSTN enable: yes
VoIP caller default DP: 1
One stage Dialing: no
VoIP users & Passwords.
User 1 ID: 11. User1 DP: 1
User 2 ID: 21 User 2 DP: 1
User 3 ID 22 User 3 DP: 1
(These are the line numbers of additional PAP2’s on our system)
PSTN to VOIP Gateway enable: yes
PSTN Caller ID none
PSTN Caller Default DP: 2
Detect PSTN long silence yes
Detect VoIP long silence yes
Detect Disconnect tone yes
VoIP answer delay 0
PSTN Answer delay 0
PSTN to VoIP gain (Set these to adjust
VoIP to PSTN gain the speech volume)
Line in Use voltage: This should be set midway between the On Hook and Off Hook voltages, which you get from the Info screen. Most public phones are 47v on hook, and 7v off hook, so the setting should be 27v. My PBX is 27v on hook, and 7v off hook, so my setting is 17v. To read this, go to the Info screen and check the Line Voltage, then go Off hook (make a call), click the reload button on your browser, and check the line voltage again.
(save these settings)
This completes the setting up of the SPA3102.
Now for the setup of the PAP2.
Open your web browser, and type the PAP2 IP into the address bar. Go to Admin, and Advanced Settings.
System tab:
DHCP no
Static IP 192.168.x.x (same sub-net as your network. Different adaptor number)
Net Mask 255.255.255.0
(save these settings)
SIP Tab: no changes.
Provisioning Tab: no Changes
Regional Tab.
Hook Flash Min & Max: change to your local settings if required.
(save these settings)
Line 1 & Line 2 Tabs.
Whether you use Line 2 depends on whether you want to have 2 phones on the PAP2. All calls from the PSTN line of the SPA3102 will go to Line 1 of the PAP2 as per Dial Plan 2 on the SPA
Line enable yes
SIP port 5060 (line 1) & 5061 (line 2)
Proxy Proxies are not used.
Register no
Make call w/o reg yes
Answer call w/o reg yes
Display name: anything you like
User ID 11 (line 1) & 12 (line 2)
(These are used to identify each line on the system)
Call waiting: no
3 way conf: no
3 way call: no
DTMF Tx method: AVT
Dial Plan: This is the dial plan I use on line 1.
(<:192.168.4.10:5061>S3|21S0<:@192.168.4.9:5060>|22S0<:@192.168.4.9:5061>)
You will have to modify it for use on other lines, or other adaptors, and the IP addresses must match your system IP addresses. Here is an explanation.
192.168.4.10:50613 All my adaptors are on subnet 4. 10 is the number of the SPA3102, and 5061 is the SIP port mapped to the PSTN line. If the handset is lifted, and no numbers are dialed the call will be transferred to the PSTN line after 3 seconds, and you will hear the outside dial tone. If within 3 seconds you dial either 21, or 22, the phone on either line 1, SIP port 5060, or Line 2, SIP port 5061, on adaptor 9 will ring. (If you only have one PAP2 then you will only need the first section of this dial plan.)
Enable IP Dialing: yes
(save these settings).
User 1 and User 2 tabs: no changes
That just about does it. All incoming calls from outside are received by the PBX, and after hours are sent to the extension connected to the SPA3102, which rings the phone on the remote PAP2 in the manager’s house. If the call is for a guest we can press the recall button (hook flash), dial the guest’s extension number, and transfer the call when they answer. As an added bonus we have a second PAP2 elsewhere on the network, and we can call between the 3 adaptors. All 3 adaptors have access to an outside line, though the PBX. I’m fairly sure it would also work through a VPN, which would mean we could take a VoIP phone anywhere in the world, and still be virtually ‘On site". I don’t know if that is a good thing or not.Hi HW,
The PBX is a Panasonic TA308. There is no special interface to the PBX, the line port on the SPA3102 is simply plugged into an extension, like another phone. Anyone calling that extension will have the call routed through the SPA & PAP2 to the remote phone.
The whole setup is totally seamless, & transparent to the user. As we are on a local network there is virtually no latency. There is a slight tendancy to echo, but the echo suppression mostly takes care of that.
THis has been a good exercise, and once I got my head around what I was trying to do, with your help, it was pretty easy. I think the hook flash timing would be the thing which gives most users a problem, as it seems to vary widely around the world. I was surprised at the difference between the US and NZ (.1 & .9 to .07 & .13). There didn't seem to be any other critical differences.
Now I am the local expert on VoIP "In the Kingdom of the blind, the one-eyed man is King." -
Issue with SPA525g registation and FXO port call calls are not disconnecting properly
Hi,
I have a UC540 and updated it to the latest IOS version with the latest firmware to my phones and i am having registration problems with SPA525g IP Phones. I updated the firmware of the phones as well and create manual tftp bindings with but still it is not registering. I run a couple of debugs (debug tftp events and debug ephone registration) I can see from the logs and in the phone that it is taking the proper VLAN and being discovered via CDP and being pointed to the TFTP server and still wont register. I can see that it is also taking its own .cnf file properly then the output sccp token regected invalid devices error is shown I have a SPA502G and it is working fine. Also there is a previous issue that all the voice port are shown as engage or offhook even the calls are disconnected thus make the main PSTN number busy am based in UAE and our service provider is etisalat I have check with them about the proper disconnection values but still it the same. That's why I have arrived in the conclusion to just update everything including the IOS and the phones firmware. I have put my config in this post, I am also trying to take the CCNA Voice exam on the 2nd week of april and I think that if i don't know how fix this issue for our customer then I would probably fail that exam. any suggestion and help is greatly appreciated cisco experts.
! Last configuration change at 13:36:42 ZP4 Thu Sep 13 2012 by Nick
! NVRAM config last updated at 13:45:41 ZP4 Thu Sep 13 2012 by Nick
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
service compress-config
service sequence-numbers
hostname UC540
boot-start-marker
boot system flash:uc500-advipservicesk9-mz.151-2.T4
boot-end-marker
logging buffered 64000
enable secret 5 $1$3CIf$.rXyHeJQrwd97X/f2dS0M1
no aaa new-model
clock timezone ZP4 4 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-3558175224
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3558175224
revocation-check none
crypto pki certificate chain TP-self-signed-3558175224
certificate self-signed 01 nvram:IOS-Self-Sig#3.cer
dot11 syslog
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.3.1 10.1.3.10
ip dhcp pool phone
network 10.1.3.0 255.255.255.0
default-router 10.1.3.1
option 150 ip 10.1.3.1
ip name-server 213.42.20.20
ip name-server 195.229.241.222
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW cuseeme
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
trunk group ALL_FX0
voice call send-alert
voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
no update-callerid
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice class dualtone-detect-params 1
freq-max-deviation 50
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 6
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 406
cadence 398 344 237 527 400
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice class cause-code 1
no-circuit
voice register global
voice hunt-group 1 parallel
list 301,302,303
timeout 24
pilot 511
voice translation-rule 4
rule 15 // //
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^9/ //
rule 3 /^0/ //
voice translation-rule 2222
voice translation-rule 3265
rule 1 /\(^..........$\)/ /9\1/
rule 2 /\(^.........$\)/ /9\1/
rule 15 /\(^ABCD$\)/ /ABCD\1/
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile INCOMING_CallerID_PROFILE
translate calling 3265
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_FXO
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC540W-FXO-K9 sn FHK143074G6
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$vjNa$OFKLhupqR8al6x2b8Xmcj/
username adminac privilege 15 secret 5 $1$NDC.$PtD0y4YGIj5SqI1gghxWE1
username Nick privilege 15 secret 5 $1$iAmL$tsg7Jf2TEND1NN.h8z2dy/
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
ip address 192.168.101.2 255.255.255.252
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport access vlan 20
spanning-tree portfast
interface FastEthernet0/1/8
switchport access vlan 100
macro description cisco-switch
interface Dot11Radio0/5/0
no ip address
shutdown
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan20
ip address 10.10.10.1 255.255.255.0
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
no ip address
ip nat inside
ip virtual-reassembly in
shutdown
interface BVI100
description $FW_INSIDE$
ip address 10.1.3.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.101.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
logging esm config
access-list 1 remark SDM_ACL Category=2
access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.3.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny ip 192.168.10.0 0.0.0.255 any
access-list 100 deny ip host 255.255.255.255 any
access-list 100 deny ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 deny ip 10.1.3.0 0.0.0.255 any
access-list 101 deny ip 192.168.10.0 0.0.0.255 any
access-list 101 deny ip 192.168.101.0 0.0.0.3 any
access-list 101 deny ip host 255.255.255.255 any
access-list 101 deny ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny ip 10.1.10.0 0.0.0.3 any
access-list 102 deny ip 10.1.3.0 0.0.0.255 any
access-list 102 deny ip 192.168.101.0 0.0.0.3 any
access-list 102 deny ip host 255.255.255.255 any
access-list 102 deny ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
access-list 102 permit ip 192.168.101.0 0.0.0.3 any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny ip 10.1.10.0 0.0.0.3 any
access-list 103 deny ip 192.168.10.0 0.0.0.255 any
access-list 103 deny ip 192.168.101.0 0.0.0.3 any
access-list 103 deny ip host 255.255.255.255 any
access-list 103 deny ip 127.0.0.0 0.255.255.255 any
access-list 103 permit ip any any
access-list 105 permit ip any any
snmp-server community public RO
tftp-server flash:/phones/521_524/cp524g-8-1-17.bin alias cp524g-8-1-17.bin
tftp-server flash:/phones/5x5/spa5x5-7-1-3c.bin alias spa5x5-7-1-3c.bin
tftp-server flash:/phones/525/spa525g-7-4-8.bin alias spa525g-7-4-8.bin
control-plane
bridge 1 route ip
bridge 100 route ip
voice-port 0/0/0
cptone GB
station-id name Cordless
station-id number 329
caller-id enable
voice-port 0/0/1
cptone AE
caller-id enable
voice-port 0/0/2
cptone AE
caller-id enable
voice-port 0/0/3
cptone AE
caller-id enable
voice-port 0/1/0
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4FXO-0/1/0-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/1
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/1-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/2
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
supervisory dualtone-detect-params 1
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/2-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/3
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/3-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.3.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec729
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/1/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/1/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/1/2
no sip-register
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 388
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 6 pots
description "catch all dial peer for BRI/PRI"
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/1/3
no sip-register
dial-peer voice 69 pots
destination-pattern 329
port 0/0/0
dial-peer voice 300 pots
trunkgroup ALL_FX0
description Local Numbers
destination-pattern 9T
forward-digits 9
dial-peer voice 301 voip
destination-pattern 2..
session target ipv4:192.168.201.2
dial-peer voice 303 pots
trunkgroup ALL_FXO
trunkgroup ALL_FX0
description **InternationalCall**
destination-pattern 88T
dial-peer voice 304 pots
trunkgroup ALL_FX0
description *EM1*
destination-pattern 9[1-9]T
forward-digits 3
dial-peer voice 302 pots
trunkgroup ALL_FX0
description **Mobiles**
destination-pattern 9.[0-9].[0-9]......
dial-peer voice 305 pots
trunkgroup ALL_FX0
description **800-**
destination-pattern 9[0-9][0-9][0-9]T
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.3.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message American Center
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.2/CCMCIP/authenticate.asp
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-4-8
load 501G spa5x5-7-1-3c
load 502G spa5x5-7-1-3c
load 504G spa5x5-7-1-3c
load 508G spa5x5-7-1-3c
load 509G spa5x5-7-1-3c
time-zone 35
date-format dd-mm-yy
voicemail 388
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh MOH2.wav
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$iDgA$MKNi2RWfsO0KjuC82kgLJ1
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
fac standard
create cnf-files version-stamp 7960 Aug 29 2012 12:00:04
line con 0
privilege level 15
logging synchronous
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
exec-timeout 0 0
logging synchronous
login local
transport input all
line vty 5 100
login local
transport input all
ntp master
end
Some of the output are not shown becaus it is to long I have attach the whole config for reference and any advice on how could I optimize and resolve my issues is greatly appreciated. ThanksNicolo - First off this stuff gets crazy sometimes. No worries about the exam. Sometimes when FXO ports go crazy it is due to battery reversal. If you go to the FXO port settings try turning battery reversal on and or off... depending on its current setting. See if that helps.
As for the 525s not registering.. These are inside the network correct? Are you connecting one directly to the UC500 with a Cat5E or Cat6 patch cable and the same thing happens? Does the MAC address on the phone match a MAC address under the EPHONE settings?
If you telnet into the UC500 can you execute a "dir" command at the CLI prompt and "CD" (change directory) into the phones folder and then the spa525g folder? Do files exist in there?
Also I only see an IP address under BVI100? This is the voice side of things what happened to the IP address under BVI1 (Data VLAN). Can you give us some information about the internal network? Cna you PING this phone system from the network? What IP address does it have? -
Setting Up VLAN and QoS for VOIP on SG200-18
We recently purchased the SG200-18 smart switch to replace a Netgear unmanaged switch. We're moving our phone service to VOIP through our local ISP as well.
I've currently got the VOIP phone plugged into Port 17 on the SG200-18 (it's a Grandstream cordless VOIP phone).
I want to put the VOIP phone on a separate VLAN from the rest of the network and optimize the QoS settings so that the VOIP phone has exceptional audio quality even during intense network traffic.
Here's my questions:
1. Do I need to adjust anything on the type of port for Port 17 (since it looks like some form of Combo port)?
2. How do I go about isolating the VOIP phone on it's own VLAN (I'm seeing VLAN and Voice VLAN settings, not sure which one to use; I tried setting a VLAN and broke Internet connectivity to the phone until I went in and removed it)?
3. Do I need to adjust any QoS settings on the switch to better optimize the VOIP phone?
A couple of additional questions about the GS200-18 in general:
1. Do I need to adjust any of the System Time Settings on the switch? I'm in Central Time.
2. Do I need to adjust any of the Green Ethernet/Energy Saving settings or should I stick with the defaults?
Also, a couple of "getting started" side questions to Cisco:
1. I've registered a My Cisco account. What do I need to do to register my switch with Cisco and associate it with my My Cisco account?
2. What are the benefits of taking out a Cisco Small Business Support Contract, and about how much would it cost on the SG200-18 (I ordered it from Provantage)? I'm curious to see if it's worth the money.
Here's my "specs":
Switch: SG200-18
VOIP phone: Grandstream DP715 and 710 expandable handsets
Plugged into: Port 17 on the SG200-18
ISP: Local ISP (Direclynx)
Connection type: 3M down/500k up DSL, moving to a wireless connection coming up which will give us faster speeds
VOIP backend provider: VOIP Innovations
Router: Apple Airport Extreme AC model (I run all Macs and iOS devices and OS X Server on the network, so using the Apple router makes setup easier, since it doesn't QoS, trying to QoS and VLAN at the switch level)
Thanks everyone!Hello,
Lots of different questions here so I'll try to make sure I don't miss anything.
1. Do I need to adjust anything on the type of port for Port 17 (since it looks like some form of Combo port)?
The way the combo ports work is you can either use the SFP slot for a fiber connection or the copper ethernet port, but not both at the same time. Other then that they just function as normal network ports.
2. How do I go about isolating the VOIP phone on it's own VLAN (I'm seeing VLAN and Voice VLAN settings, not sure which one to use; I tried setting a VLAN and broke Internet connectivity to the phone until I went in and removed it)?
It sounds like you created the VLAN correctly and assigned the phone, however there wasn't anything doing any routing for that VLAN. You would need to have a VLAN capable router or a layer 3 switch so that something would act as the default gateway for the voice VLAN and route the traffic for you. Since there was nothing like this your phone lost it's connectivity to the internet when you placed it in the new VLAN. I don't think the Airport is VLAN capable, but we will come back to that.
3. Do I need to adjust any QoS settings on the switch to better optimize the VOIP phone?
Once you have a seperate VLAN setup for the phone properly you only have to tell the switch what your Auto Voice VLAN is going to be and it will automatically apply recommended QoS settings for the Voice VLAN and prioritize the voice traffic. There are ways to do this manually and even with the phone in the same VLAN however the are considerably more complicated.
1. Do I need to adjust any of the System Time Settings on the switch? I'm in Central Time.
The system time isn't always very important. You can set the correct time zone, however you should know the switch does not have a battery in it to keep track of time, so if/when it reboots or loses power the clock will reset. If you would like the switch to maintain accurate time you should setup an NTP server so the time is automatically updated from the internet. The switch will keep your timezone settings once you save them. Time is mostly important for logging and things like that, so you can configure it if you like but it is not necessary.
2. Do I need to adjust any of the Green Ethernet/Energy Saving settings or should I stick with the defaults?
Green ethernet simply reduces the power usage of the switch slightly, so unless you are having odd issues where ports are disconnecting, I would just leave them at the defaults.
1. I've registered a My Cisco account. What do I need to do to register my switch with Cisco and associate it with my My Cisco account?
There isn't really a way to associate your Small Business devices with your Cisco account. If you ever call in for technical support we will use your Cisco account and your serial number to create a support case, but even then they aren't linked together. If you decide to buy a support contract, that will be linked to your switch's S/N and your Cisco ID, so in a way that would associate them together. Devices being associated with Cisco accounts is something more common with Enterprise equipment, and mainly has to do with technical support cases.
2. What are the benefits of taking out a Cisco Small Business Support Contract, and about how much would it cost on the SG200-18 (I ordered it from Provantage)? I'm curious to see if it's worth the money.
There are a few advantages to a Support Contact. Your switch comes with a Limited Lifetime warranty that includes 1 year of technical support and return to factory hardware. With a service contract you get 3 years of technical support and next business day Advanced Replacement of the switch if it need to be replaced. I just did a quick google search, and it looks like a contract (part #CON-SBS-SVC2) costs about $50.
So there are a few other things to consider however.
As a frame of reference the average VOIP call uses about 64 - 128 kbps max.
Since you don't have a VLAN capable router or a layer 3 switch, a separate voice VLAN may not be an option. You also mention that the Apple Airport does not do QoS, meaning we will only be prioritizing the voice traffic while it is on the switch. When it is passed off to the Airport to be routed out to the internet all of the QoS settings will be lost, and normal network traffic will get the same priority as voice, since that is all up to the Airport.
With one phone the hassle of getting more equipment and setting up advanced QoS isn't really worth it, especially if the link to the internet isn't going to be participating in QoS.
One last thing I wanted to mention is you are switching to a wireless internet connection. I would ask them how their latency and jitter is, as these two network statistics greatly effect voice quality, and usually wireless performs worse when it comes to voice traffic.
I hope this information helps, if you have any more questions just let me know.
Thank you for choosing Cisco,
Christopher Ebert - Network Support Engineer
Cisco Small Business Support Center -
I have E65 with latest software (2.0633.65.01) and there is problem with WLAN and VoIP connectivity.
My VoIP connection work's fine, but when i disconnect VoIP service, WLAN connection stays active (i see WLAN icon). If i try to disconnect manually through Conn. mgr. WLAN reconnects after few seconds.
VoIP is set up to connect 'When needed', WLAN Show availability is 'Never'.
The only way to disconnect is to switch the phone off and on.
Maybe somebody have solved this annoying behavior?If you're seeing a symbol with two arrows - one pointing left and the other right, that is not the WiFi symbol - that is the 3G connection symbol. Disconnecting wireless won't help in that case - it's using your provider's 3G data connection. The WiFi log is like a radio tower with "waves" on either side. To stop the network connection, you should find out which applications are requiring a connection to the network, and stop those from running/updating automatically (e.g. checking email every few minutes).
Now about why the 5800XM doesn't see your home router easily - there could be several issues. I would suggest posting complete details about your router (some people have posted issues with specific routers) and doing some basic test like standing right next to the router and checking to see if your 5800 can see it.
If you want to start fresh, just go to your Connectivity > Destinations, and delete the entry for your home WiFi connection and start again. However, this will work only when you're not connected, so you may want to turn off your phone, then turn it on again and do it while it's not connected.
Lumia 920, Lumia 800
Nokia N8-00 (NAM, Product Code: 059C8T6), Symbian Belle, Type RM-596, 111.030.0609
Nokia 5800 XpressMusic (NAM, Product Code: 0577454) Software v51.2.007, Type RM-428
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