VoIP on FR Issue

I have VoIP installation, with 40 branches linked to head office via FR start-topology PVC links each with 64K CIR.
Each branch having 2 analog phones attaches to branch router 1751-V.
Voice trafic for each branch has given priority and has allocated 48K bandwidth for voice on each PVC.
Some branches experiencing intermittant voice quality problems.
Q1. Suppose if a branch link is having two on-going voice conversations. If a third call trie to enter the link can it cause the voice quality problem bencause the link cannot handle 3 calls ?
Q2. Will installing a voice gateway to do call admission controll at the head office will solve the problem ?

Answer to your Q1 is yes. The third call will affect the two active simultaneous calls. You should make sure the priority you have defined is enough for the active calls that you expect simultaneouosly.
Check the following URL:
http://www.cisco.com/en/US/products/sw/iosswrel/ps1830/products_feature_guide09186a0080087b13.html
For CAC you can either use
1. RSVP or
2. Gatekeeper solution
I hope that helps
Javed

Similar Messages

VOIP Dial out issues with SRP547W

Hi All, this is my first post so I appologise if I have posted it in the wrong place up front.
I have recently purchased an SRP 547w replacing an 877w and linksys pap2t voip box.
I have 1 major issue at present - no matter which number I dial, I cannot ring out. I get what appears to be an engaged tone.
History: I used to run the linksys papt2 behind the 877w and it worked without issue. Ports opened and forwarded were:
udp 5060-5061
udp 8000-8001
my SIP port is 5060
the RTP ports are 8000-8001
I know this works on the SRP547 as without them, I cannot make any calls into the system.
These settings worked perfectly on the PAP2T; I have copied them across to the SRP547W and opened the same ports. I can call VOIP device on the specified number, but still no dialing out.
The 2 main differences I can see on the new device 1) The VOIP is integrated into the router so I'm not routing traffic in exaxtly the same way as I was with the PAPt2 and the second being that the SRP547W also has the ability to connect a PSTN line - which I do not have.
I wonder if this has anything to do with the issue?
For the record, I have diabled the firewall - same result; additionally, if I connect the PAP2T to the SRP547 and forward the ports (or enable UPNP) then I can actually dial out which means I think its safew to say that my issue is within the SRP547W.
I can conrim that the SRP547 does 'hear' the tones from the phone. When I dial certain numbers (to force certein codecs) the system responds to them.
If anyone else has had a similar issue, could you help me out with some direction?
I'd post logs, but I'm not sure how to get them off this device (yet) - I've only had this device for 4 days.....

I forgot to post an update.
Get this, I figured out the problem and it was Sun Rocket.
They did a firmware update over night that caused it to break.
After about three months of working with Linksys, Comcast and Microsoft and tyring everything I was told. It still failed.
Then after another month, I finally was contacted by an engineer from Sun Rocket who was giving me advice over everything we already tried.
I asked him if there was a firmware update from the date I put the complaint ticket in. He checked the database and sure enought, that is what it was.
Their firmware update did not work well with the Linksys router. He rolled back the firmware and now it works perfect.
The SunRocket tech said he put me on a list of Sun Rocket Gizmos NOT TO update firmware.
Well , about 2 months later, they updated it by mistake and now I am going through the pains with SunRocket all over again.
I told them how they fixed it but the Level 1, 2 and 3 techs don't seem to know how to roll back the firmware and they did not have any notes from the engineer on what version to roll it back to.
The the root cause was SunRocket's Gizmo.
Trust me, I've tried chaning the MTU to remoting in to the Sunrocket Gizmo and change the bandwith settings ect ect. I've been through the ringer and I still have problems with them.
The SunRocket Gizmo can not handle the high bandwidth like the linksys router can.
Good Luck, I only have SunRocket for another month and I am thnking about going with Vonage or Comcast.
Here is a picture of the SunRocket Gizmo I am using.
Message Edited by TheInfamousOne on 01-05-200709:56 PM
Message Edited by TheInfamousOne on 01-05-200710:00 PM

Voip(gizmo & truphone) issues...need correct setti...

have been trying since i had the n95 to connect via voip/internet telephone been using gizmo...recently installed truphone and got the same result....unable to connect...so i figured im doing something incorrectly
can someone tell me the settings for these services(gizmo and truphone)
i cant seem to get voip/internet telephone configured no matter what i tried...

The "Always On" option is not recommended (atleast by me)
Create Your Sip Settings. (When Needed)
And Make a New Internet Tel connection
Choose ur Sip profile
Use the Internet Telephone applet to connect
Use the same applet to disconnect
You can have as many sip profiles and internet tel connections as you prefer.
I use to connect 7 providers at a time.
You can use the same Internet Telephone applet to change the active provider (to call)
You can receive at any of the connections at any time when connected.
The steps are:
1. Create a SIP Profile (Tools->Settings->Connection->SIP Settings on N80)
2. Create a Internet Tel Profile (Tools->Settings->Connection->Internet Tel Settings on N80)
3. Use Internet Tel applet (Internet->Internet Tel. on N80) for connection and disconnection
If you add a shortcut of the Internet Tel to your Active Standby, with a single click you can connect and disconnect. But you cannot change the active connection.
Hope this helps
N80 IE, E51, E71

Voice Call issues

Hello,
We have Exchange 2010 and Blackberry 5 in our environment. This query is for a single user(User A)only who is having exchange active sync and Blackberry.
When users call him via Blackberry address book, the call lands to user B instead of User A.
If the user is called directly from voip, landline the issue does not occur.
The issue is observed only if any user calls this user A to his blackberry. However, the issue is not observed on I phone. The phone has been wiped alreday.
Regards
Ajit

Can you share the SIP dialog between CUCM and the MCU? INVITE, 100, 180/183, 200 OK, ACK?
CUCM can do the packet capture directly from the CLI as long as you run it from whatever node the SIP trunk is being sourced from (primary of that trunk's CMG).

VOD Issues

Having issues with our VOD on all three TVs. The picture and sound is choppy. It is so bad that you can't watch anything. I looked online and Verizon said that VoIP can cause issues, so I disconnected it, reset the router, but still have the problem. I'm pretty sure it has something to do with the router, but not sure how to fix it.
Thanks.

I have tried everything I've seen listed on the boards short of replacing my boxes. I have unplugged all the boxes in the house, unplugged the cable lines and reattached (to ensure proper fit), I unplugged and reset the router, then plugged all the boxes back in. Bubkis. Someone suggested that VoIP can cause issues, so I unplugged that, rebooted the router, still nothing. I am very frustrated that it doesn't work and we can't watch anything on VOD. I'm also frustrated that we have continual issues with our boxes (and they are all different - standard, HD, and HD DVR). Someone please help us!
PS. theparanorm, any relation to paranorman?

VOIP phone not getting IP from correct VLAN

Hello,
(sorry not sure if it is a VOIP or Switch issue)
We are installing a couple of VOIP phones in a new office, the VOIP service is hosted and the phones are Yealink. Off these VOIP phones are PC, but both the phone and PC get an IP from the DATA VLAN, I want the VOIP phone to go on it's VOICE VLAN.
Here is an example of the port, it is a Layer 2 switch (2960x):
description PC & IP Phone
switchport trunk native vlan 10
switchport trunk allowed vlan 10,62
switchport mode trunk
spanning-tree portfast trunk
spanning-tree bpduguard enable
The trunk port:
interface GigabitEthernet1/0/24
description ROUTER-LAN
switchport trunk allowed vlan 1,10,62
switchport mode trunk
speed 100
duplex full
spanning-tree portfast trunk
Router:
ip dhcp pool Data
   domain-name gb.vo.local
   default-router 172.30.12.8
   dns-server 192.168.21.10 192.168.21.11
   option 66 ip 192.168.28.89
ip dhcp pool Voice
   network 172.30.212.0 255.255.255.0
   domain-name gb.vo.local
   dns-server 192.168.21.10 192.168.21.11
   default-router 172.30.212.1
interface FastEthernet0/1
description LAN
no ip address
duplex full
speed 100
interface FastEthernet0/1.10
description DATA LAN
encapsulation dot1Q 10
ip address 172.30.12.11 255.255.255.0
standby 1 ip 172.30.12.8
standby 1 priority 110
standby 1 preempt
standby 1 track 1 decrement 10
interface FastEthernet0/1.62
description VOICE VLAN
encapsulation dot1Q 62
ip address 172.30.212.1 255.255.255.0
Does it all look ok?
I can see the phones:
sh lldp neighbors
Capability codes:
    (R) Router, (B) Bridge, (T) Telephone, (C) DOCSIS Cable Device
    (W) WLAN Access Point, (P) Repeater, (S) Station, (O) Other
Device ID           Local Intf     Hold-time Capability      Port ID
IPPHONE             Gi1/0/14       138        B,T             0015.654c.6630
IPPHONE             Gi1/0/19       156        B,T             0015.654c.65f6
Thanks

All was all good, it was jus tthe network command missing:
ip dhcp pool Data
network 172.30.12.0 255.255.255.0
   domain-name gb.vo.local
   default-router 172.30.12.8
   dns-server 192.168.21.10 192.168.21.11
ip dhcp pool Voice
   network 172.30.212.0 255.255.255.0
   domain-name gb.vo.local
   dns-server 192.168.21.10 192.168.21.11
   default-router 172.30.212.1

WRT54G; What is latest version of firmware ?

I have a WRT54G v4 and somehow I had noticed I upgraded to 4.30.5. When I went to web to see if I still ahd latest it said 4.21.1 was latest. So, I quickly re-flashed with 4.21.1. I went into the 4.30.5 release notes (below) and it DOES say WRT54G. What's up ?
Linksys, A division of Cisco Systems, Inc.
Product:                WRT54G
Classification:         Firmware Release History
Firmware Date:  04/27/2006
Release Date:           05/25/2006
Last Firmware Version: 4.30.5
Firmware 4.30.5
- Resolves UPnP vulnerability reported by SANE
Firmware 4.30.2
- Resolves issue with filter name
- Resolves issue where service name for PPPoE cannot be added
- Resolves issue with detecting UHP modems
- Resolves issue where DynDNS Custom and Static accounts cannot be used
- Resolves issue with WEP passphrase not recognizing spaces
Firmware 4.20.7
- Resolved issue with SecureEasySetup button delay
- Resolved issue with SecureEasySetup overwritting current wireless settings
- Resolved issue with not being able to disable SecureEasySetup button
- Resolved issue with Diag light being constantly lit after upgrading to 4.20.6 on v.1 hardware
- Resolved issue with iDefense security vulnerability
- Resolved issue with L2TP disconnect
- Resolved issue with SecureEastsetup long button push not resetting wireless settings
Firmware 4.20.6
- Resolve issues with setup wizard
- Updated wireless security menu
- Resolve issues with port based QoS
Firmware 4.00.7
- Adds SecureEasySetup push button support
- Resolves large file transfer issues
- Resolves issue with enabling TKIP after enabling WEP
- Updated QoS features
- Resolves issues with multiple Access Restrictions policies
- Resolves issue where multicast breaks when MAC filter status changes
Firmware 3.03.6
- Supports hardware version 3
- Updated help file
Firmware 3.03.1
- Wi-Fi certified
Firmware 3.01.3
- Updated wireless driver
- Supports hardware version 2.2 (cannot downgrade to previous versions)
- Resolves issue with VoIP adapters
- Resolves issue with long domain names
Firmware 2.04.4
- Updated wireless driver
- Updated certificate generation for https access
Firmware 2.04.3
- Resolves issue with disabling firewall
- Added QoS function
- Added Port Triggering function
- Added L2TP option for WAN connection type
- Added ability to back up and restore configuration files
- Added Wireless isolation function
- Added ability to filter internal NAT redirection
- Added ability to restrict access to web GUI from wireless clients
- Added ability to filter IDENT port 113
- Resolves issue with HeartBeat WAN connection
- Added support for 40/56bit cipher to support old browsers to use https
- Resolves issue with PPTP passthrough
- Changed WPA Group Key renewal time range to 600~7200 seconds
- Added IGMP proxy support
- Added ability to allow web GUI access only through https or http
Firmware 2.02.2
- Updated wireless driver to support all versions of WRT54G hardware
- Adds support to allow WAP54G connect as a repeater
- CTS protection mode set to disable by default to improve wireless performance in normal environments
- Resolved issue where the WLAN LED stays on even when wireless is disabled
- Resolved security vulnerability causing the web server in the router to crash
Firmware 2.00.8 (WRT54G v.2 only)
- Updated wireless driver
Firmware Version 1.42.2
- Updated wireless driver to 3.31.15
- Added URL and contents filters
- Added L2TP passthrough
- Added HeartBeat Signal support
- Resolved issue with Connect on Demand
Firmware Version 1.41.2
- Updated GUI format and layout
Firmware Version 1.30.7
- Adds UI control to enable/disable frame bursting
- Resolves issue with firmware upgrade error in Internet Explorer 6.0.2800.1106.xpsp2.030422-1633, on Windows XP pro.
- Send 192.168.1.1 to disable DynDNS
- Resolves issue with using the 4th tx key for WEP
- Resolves issue with disconnecting PPPoE causing WPA to be disabled
Firmware Version 1.30.1
- Adjust PPTP max MTU value from 1400 to 1460
- The router is now certifiable for the UPnP logo
- Complies with final 802.11g standard
- Improved performance asymmetry between the transmit and receive paths
- Receive Broadcast packets from renew state. Some DHCP server (Dlink 300G+) send broadcast ACK after it receives a REQUEST
- Fix: The firmware upgrade will fail, if DDNS updates while upgrading
- Fix: If first DNS address fails, the second or third DNS address will be used to resolve domain names
- Added the following items under Setup Page:
    - Router Name
    - Manual Time Server
    - MAC Clone
    - wireless domain
    - Security Settings
- Added the following items under System Page
    - UPnP enable/disable
    - Ping Test and Traceroute
- Added the following items under Wireless Page
    - CTS Protection Mode
    - Basic Rate
- Added the following items under Status Page
    - Router Name
    - Start IP Address
    - End IP Address
- Added Wi-Fi Protected Access(TM) support including AES and TKIP encryption
- Resolves issue with Port Forwarding name showing up as garbage characters
- Added "B-Only" option to wireless modes
- Added "Default" option to Basic Rate settings
Firmware Version 1.02.1
- Incorporates updates found in the Draft IEEE 802.11g version 6.1 specification
- Resolves interoperability issues with legacy 802.11b devices under Mixed Mode
- Resolves a performance bug where the performance in a mixed b, g network did not go up once the last b client disassociated
- Resolves issue where WEP degrades performance
- Resolves issue with TZO DDNS Service
- "Domain Name" replaces "name"
- TFTP server will only verify image file name "xxxx.bin"
- MAC Address Filter under the Advanced tab works on both wired and wireless clients
- Fixed a bug that prevented wireless clients from connecting to the Internet if there was no active wired PC (PPoE customers in Europe and China only)
- Support for UPnP enabled applications such as MSN Messenger
- Fixed a bug that caused computers connected to the LAN ports to be dropped at some interval only to be immediately re-connected
- Fixed Dynamic Routing issue between 2 Routers
- Allows User to name Static Route Configurations
- Modified PC List Web Page to avoid confusion
- Stabilized Wireless-G to LAN and WAN Throughput
- Stabilized Wireless-B to LAN and WAN Throughput
- Improved LAN-WAN Throughput
Firmware Ver.: 1.01
- Initial release

History.
Linksys released the 4.30 firmwares for the WRT54GL model which are compatible with the WRT54Gv1-v4.
I guess some people were confused to see WRT54GL on the web interface of the router although they had a WRT54G.
Thus they re-released the firmware for the WRT54Gv1-v4 with 4.21.* showing the correct model name.
You should be able to use either firmware for either router. If you check the release notes of 4.21.* and 4.30.* you'll notice that they basically have the identical history of changes. 4.30 is now a little bit more advanced if you look on the WRT54GL downloads page.

Callto:// hyperlink in Numbers (and Excel)

I have a spreadsheet table that includes phone numbers.
I would like to create an extra column with hyperlinks that when clicking on them will automatically use my prefered VOIP application.
Issue 1:
When I use the callto:// hyperlink in Numbers '09 (2009 when was that again) it doesnt do anything. After that I can't create any hyperlink at all anymore (even after closing and restarting numbers) what is the prober way to get to do this?
1a. In Excel 2011 it automatically adds file:// in front of the callto:// which of course does not work
Any suggestions on how to add other then http://, file:// or mailto:// hyperlinks in excel 2011
Issue 2:
when typing callto:// directly in the browser, to goes to Skype. How to change the default program used for the callto:// statement?
Thanks in advance for you help.

Hyperlinks are limited to http and mailto. Excel has the additional ability to link to files. You will not be able to create callto hyperlinks.
Your question about how to select your default callto app might not get answered here in the Numbers forum. You might want to ask that in a different forum that has a wider audience. Or maybe an Internet search on callto to see what parameters can be specified.

SPA3000: Lowest MTU size

I can let my DSL-modem lower the MTU size as soon as I receive of make a call.
This is done with IPQoS EF Timer.
I want to do this so other heavy traffic like P2P-clients get slowed down when I am calling. I set MTU at 512 but my call gets disconnected after about 25 secs. when I set it at 768 my call doesn't get disconnected.
I would like to know the official lowest MTU size the SPA3000 can work with.
I add to this that it's not a VoIP Service Provider issue as I made a call with another ATA, same MTU setting (512), and the call didn't get disconnected.
SPA-3000
2.0.1(5673)
3.1.20(GW)
Thanks in advance.
Message Edited by caesu on 11-26-2007 12:40 AM

I will try to rephrase what I already have said.
First, for the UDP datagrams, fragmentation is not supported.
If you have to send 501 bytes UDP datagram, with MTU=500 it will get dropped. No excuse. No fragmentation. Packet is lost. If it's important SIP packet to keep the "session-in-progress" this may cause session drop.
Second, the SIP UDP datagrams can be large. As it's plain text messages, there is no compression / optimization on those datagrams. Within the SIP language boundaries, SIP device is free to compose the SIP UDP datagrams as needed. I have seen logs/debugs with the SIP UDP datagrams close to 1300 bytes.
Third, SIP UDP datagrams are sent moreless asynchronous comparing to RTP stream. At the beginning of the call, device(s) and VoIP/SIP switchboard will use the SIP dialog to initiate the phone call. After this, the RTP UDP datagrams stream is exchanged moreless independently on the RTP port(s), while device(s) continue to speak with switchboard with the SIP messages. Those messages may bring info like : ringing, call connected, call ended, call waiting on the line, invitation to another call / conference.
Fourth, for priorizing the RTP datagrams over other traffic, the RTP packets are (can be) marked by SIP VoIP device with certain TOS type. This TOS type can be used by QOS-enabled routers to pass the RTP stream first, SIP second (different TOS) and other traffic last. Using MTU for this purpose which MTU is NOT intended to may cause you unpredictable troubles anytime.
Fifth, I realy doubt you can configure SPA-3000's MTU size. I did not find it in the configuration. So very likely, SPA-3000 sends all packets as needed, but your MTU-limited device does not transfer the longer datagrams further, as fragmentation not possible. As UDP works without ACK mechanism on IP level, the SPA-3000 device will never notice the datagrams are dropped on the transport path.
You shall collect the SIP debugs (SPA-3000 can generate them to syslog server) and possibly the capture of the network traffic (for example using wireshark) between the SPA-3000 and your MTU-limited device (using port mirroring and/or HUB to ensure all packets are sent to all cards).
Here, compare the SIP debugs and SIP datagrams in the wireshark to determine, how big the SIP UDP datagrams are, and which SIP message is realy not transferred. You may collect two sets, one with standard MTU, one with limited MTU and you shall get your answer how big the SIP message realy is.

CERTIFCATE

How do I remove the following mesaage which comes up alomost very time I wish to check messages. I presume I have a corrupted vertificate!
You are attempting to open a secure connection, but the server’s certificate chain is invalid
Options:=
Continue
Close Connection
View Certificate
Trust certificate
View: =
X*.voip.blackberry.com
Issuer: =
? Thawtw SSL CA
John Denley

Any chance these are FAcebook messages you're opening?
"You are attempting to open a secure connection to fbcdn-photos-a.akamaihd.net" appears when logging...
If it's just email messages, you might look and see if that message has a "don't show this again" check box?
1. If any post helps you please click the below the post(s) that helped you.
2. Please resolve your thread by marking the post "Solution?" which solved it for you!
3. Install free BlackBerry Protect today for backups of contacts and data.
4. Guide to Unlocking your BlackBerry & Unlock Codes
Join our BBM Channels (Beta)
BlackBerry Support Forums Channel
PIN: C0001B7B4 Display/Scan Bar Code
Knowledge Base Updates
PIN: C0005A9AA Display/Scan Bar Code

ISA550 Series and Softphones

All,
I have a ISA570 appliance installed with a combination of Physical VOIP handsets and softphones.
We run a call center so we have wo different VOIP providers, redundant WAN, etc.
Primary VOIP Provider (Physical and Soft Phones) - All working order
Backup VOIP Provider (Softphones only) - Works, but painfully slow
According to the Provider, they see the phones try to register and all seems well, however when a call is placed to or from a softphone, it takes approximately 30 seconds to connect.
Hooked softphone laptop direct to other ISP router and all works fine.
So, it is in the ISA570
Created a VLAN to make it easier to work.
Created rules for WAN -> VLAN of VOIP
Created rules for VLAN of VOIP -> WAN
Same issue.
Provider says not to use ALG SIP.
Any thoughts?
Brandon

Yes, all of the devices use the primary WAN1.
1. We use company A and with them have physical phones and laptop softphones that all work fine.
The physical phones run through one switch and are all on a vlan together.
The laptop softphones run on our standard vlan (since they are in the computers) and all work fine.
There has been no special setup for either.
2. We use company B and with them only have softphones on the same exact laptops as Company A.
The softphones are using the same softphone program on the same laptops as Company A.
They do work, only they take 30 seconds to register (and sometimes wont register) and 30 seconds to connect a call. As you can imagine we do not think this is a going to be a good backup option but the company insist that their product is fine and it is just our configuration in our ISA. And I tend to agree as it works, just super slow like the ISA is not happy with some request, but then ultimately agrees to let the traffic through.
Indeed, we connected to a third, independent DSL connection here at the office (that is in no way connected to the ISA or our network) and it connected and made calls immediately. So, we do think it must be some policy I have not configured correctly.
In reference to the the separate VLAN. I made a VLAN called VOCAL for a laptop to be connected to so that as I made changes to that VLAN/ZONE I could not mess up daily activities of the office.
I have not yet configured and QOS.
Brandon

E3000 issue with VOIP caller ID

Hi,
I have a E3000 with newest firmware that has a VOIP by Velcom. It's a Grand stream HT–286.
My device is working perfectly with the router. Can call and recieve calls. the caller ID (Name and number that appears is accurate). The only problem is the caller ID's time and date are always wrong.
When I unplug device, it goes back to Jan 1 and time 12:00am.
Any call I then get is wrong.
In the router, I've enabled SIP ALG. And the firewall is only filtering ident (port 113) and Anonymous Internet requests.
The device has a hard coded internal IP (192.168.123.110). and I reserve the IP using mac filtering.
The device has an administration page that is unavailable to users as it's locked by a password. On this page has an NTP server setting that the de vice uses to get it's time/date.I see on the information page of the device that iot's registered to a sip server.
The company is saying that it's a firewall issue. I'm not too sure how to proceed. I've searched the forums and haven't found anyone with this specific issue.
Anyone has had this problem before can could point me in the right direction?
Thank you.
Solved!
Go to Solution.

It's difficult to tell without knowing the configuration. But from what you write your voip device doesn't know the correct time. Thus, either it is supposed to use NTP but the NTP server isn't reachable or it is not even using NTP and the time is supposed to be set manually in the device. Maybe it's the latter and it has "lost" its time because of low running buffer battery in the device which keeps the clock going while switched off...
Whatever it is, so far is sounds to me like a problem with the device. Only if the router blocked access to that NTP server it would have been caused by the router. But I kind of doubt that...

About the voip issue being not available(not a sol...

Hi Friends
I have formatted my phone.Reinstalled the firmware and installed sip_voip3.0 settings v1 (yes v1) With v2 it was bouncing me of the menu but this time the menu was available.I did the settings and i saw that my phone was connected.There was a globe sign at standby screen.Everything seemed perfect until i tried my first call.The phone says "net calls enabled" but they are not..I can not call anywhere and also i cant be called.It does not work.Maybe someone may continue and find a solution from here...Just wanted to share with you all suffering from voip issue.
Thanks

yes this worked for me - I was able to place a SIP Call using 'SIP_VoIP_3_0_Settings_v1_01_en.sis'
Available at:
http://www.forum.nokia.com/info/sw.nokia.com/id/105455c9-654b-427f-99c7-202aac27a8d8/SIP_VoIP_3_0_Se...
The funny thing i was going to try this earlier - but I was too lazy to sign up an account. The only thing that sucks is there is no way to start or stop the service with the 'internet tel' so once it is running it is running...
Well guess this proves Nokia was lying - should have done this LONG ago.
One note if you have a + at the start of your contacts phone number you might have issues.
I love my purple screen, 1/2 day battery, sketchy touch screen and $800 price tag - I call it my Lumia 900

Unable to issue auto qos voip on CAT2950SX-24 using IOS c2950-i6q4l2-mz.121

Hi,
I am unable to issue the command auto qos voip on the interface fa0/1. I am also not able to see the command using the "?". But based on the Cisco documentation, the IOS is able to support that command. Any can help or have any ideas? Here is the documentation link
http://www.cisco.com/en/US/products/hw/switches/ps628/products_command_reference_chapter09186a00804761fd.html#wp4214415
Thanks

Hi
you can use the below link to find out whether the image installed in your comes with the support for auto qos or not.
http://tools.cisco.com/ITDIT/CFN/jsp/index.jsp
regds

VOIP Sound Issues

Hi I have a new MacBook Pro and am experiencing terrible sound issues with both iChat and my VOIP. I use a belkin router for the Linksys modem from Verizon.
1. Will speakers help with the iChat Sound
2. What router do you recommend for VOIP. Will Airport Extreme work?
3. Any tips for getting this working right?

Please describe your sound issues in more detail. What exactly do you hear or don't you hear?
Could it be a right speaker distortion as described in this thread?
http://discussions.apple.com/thread.jspa?threadID=388368&tstart=0

VoIP on FR Issue

Similar Messages

Maybe you are looking for