Skype DECT telephone (e.g. Philips VOIP 855) and n...

Hi,
I can't find any info on this - maybe someone knows: Can I continue to use my Skype-account on the computer (e.g. for instant messaging) even though I use a telephone such as the Philips VOIP 855? Or do I need a separate account just for the hardware telephone? Obviously I won't make calls at the same time through the computer and through the telephone...
Thanks for any answer to this (obvious?) question.

Yes, you can run them both on your phone and computer/mobile no problem.

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Philips VOIP 855 - SkypeOut call connects but not ringing on other side

I've got a Philips VOIP 855 and Skype worldwide subscription. It has been working fine for a few years but since last week I have been unable to make SkypeOut calls. I have tried:- Replacing the base unit, charging unit, handset and cables with a second set I have.- Disabling the firewall on internet router- Powering off and on- Calling in via the landline works OK- Calling out via the landline works OK- Calling in via Skype-to-Skype works OK- Calling out via Skype-to-Skype works OK- Calling out via Skype-to-landline/mobile rarely works. Sometimes it connects as usual (1/10 times). The other 9/10 times, the call appears to connect and I can hear the ringing tone but then it disconnects itself with beep-beep-beep sound after about 20-30 seconds. When I ring the other side with my mobile they say their phone never rang. I've confirmed this when I call my own mobile. I can't figure out why it works on occassions but not the majority of the time. Of course, Skype status says online when I make the call and when I view my account history on www.skype.com, it lists all my dialed numbers with a duration of 00:00:00. I've tested the Skype account from my mobile and it appears to work fine. Anyone have any ideas?

Hello,
We have asked our PSTN team to investigate further.
Could you please specify from what number to what S2G number you are calling?
Thanks.
TIME ZONE - US EASTERN. LOCATION - PHILADELPHIA, PA, USA.
I recommend that you always run the latest Skype version: Windows & Mac
If my advice helped to fix your issue please mark it as a solution to help others.
Please note that I generally don't respond to unsolicited Private Messages. Thank you.

Philips VOIP phone install won't run on Win 7

Hi -
I've got a Philips VOIP 0801B/37 phone. It has an install disk. When I run the install disk it says it only runs on Windows 2000 and Windows XP.
What I want is the microphone. What does it take to get a bigger selection of microphones in the Skype dropdown list of microphones? All that shows for me is a single microphone for any of my headsets (a Realtek microphone). The Echo Sound Test Service does not record any messages when I try it.
Doug

Installing the standalone version just postpones the problem until you install the cloud version, again.
The SLCache folder has bad data in it that confused LR 5.5 and again LR 5.6 until it was cleared, via Solution 4 on this page. Of course there could be other issues, so trying each of the solutions in order could be useful, but Solution 4 is the one that finally worked for me, with LR 5.5 on my home computer, and LR 5.6 on my work computer, where I'd installed the standalone 5.5 to bypass the issue and it came back with LR 5.6:
Lightroom doesn't launch or returns "Develop module is disabled" error after 5.5 update
Once you remove the contents of the SLCache folder other Adobe programs will ask to be relicensed like Elements via a serial number or the cloud programs by logging in, but that is just once after you clear it not each time.

GE - SKYPE - Dect 6.0 Cordless phone doesn't compl...

Hi,
Two days ago, my skype phone model "GE - SKYPE - Dect 6.0 Cordless phone" stopped to complete calls. Weirdly skype (PC version) was with the same problem that day.
The call rings but when the other side answer, both sides can't hear nothing.
Today PC version is working normally however GE phone doesn't work anymore. I have tried other land numbers, but it doesn't work, people get the phone, but we can't hear each other.
I'm wondering if this kind of cordless skype phone has a expiration date of usage, even if is an old phone I believe that Skype should give support for this products.
Coincidence or not, Skype has just announced a new GE cordless phone (GE Digital Cordless Expandable Telephone)
I'm posing here because I can't find any other place to contact skype.

Hi jcantos,
Please be assured that your GE Phone is covered under a 12 month warranty and if the unit is deemed as defective, eBuyNow will reship them a new unit at no cost.
Before taking further steps please allow us to give you this troubleshooting to see if we can help you with your problem. Please ensure that the handset is within range of and linked to the base station. This is necessary for the telephone to charge.
When you put the handset into the charging cradle, you should see the following:
1. Charge indicator (RED LED) will light up permanently until the handset is lifted out of the charging cradle.
2. Handset initially displays "charging" within several seconds when the battery is low.
3. Handset will display scrolling battery icon for charging.
Please let us know the following:
1. Is the RED LED always off? Check if the handset charge pin really contacts to the charge cradle spring, as well as to verify if the sub-adapter connection is loose or not.
2. Is the battery not installed well? The handset will display "no battery". Disconnect and re-connect the battery pack into your phone.
3. Does the handset display "charging"? Wait a few minutes. If the handset starts to enter into stand by mode with the scrolling battery icon, the handset can be charged up.
4. Does the battery icon on the phone indicate battery low, but when put in the charger, indicates battery is fully charged?
5. Have you managed to successfully make a call from the handset?
6. Is the indicator light on of your base station?
The more information you can provide us with, the better we can assist you.
If you already went over the above and are still having problems, please contact our eBuyNow's Support team by visiting ebuynow.info/en/contact/ (add the http:// as this forum does not allow hyperlinks) and one of our agents will gladly assist you.
Best,
eBuyNow.Support.Pro

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!

Hi, i have a macbook air and i've been having problems with the camera when i'm using skype. i know im no the 1st one and i'd like to know when apple or someone 'll do something about this.

hi, i have a macbook air and i've been having problems with the camera when i'm using skype. i know im no the 1st one and i'd like to know when apple or someone 'll do something about this.

Try reinstalling Combo Update.
http://support.apple.com/kb/DL1676
Best.

Skype number is suddenly not working for me and my...

Skype number is suddenly not working for me and my colleagues to RECEIVE outside calls. Any ideas?
When I try to call my skype from my cell, it doesnt ring on either end. It appears connected on the cell phone since the counter clock for the call is increasing, but it doesnt ring or connect.
This is the same for me and at least 2 of my colleagues. It was working all day, till suddenly it did not.
When outside people call, it just gives us a missed call or voicemail email, but does not show a missed call on our skype.
When I call an outside number from skype, it connects.
Thanks for any help ASAP!

I think this might have been happening to me earlier. If you wait like 30 seconds when you called a machine said "All circuits are busy. da da da". I was looking for an answer and google only brought stuff from like 2010. It's working now after an hour or so.
Sucks cause I use skype for work and was desperate to find a quick reponse but couldn't.

Asha 501: VoIP, TLS and certificates

Hi
I've recently added sip account to my Asha 501. It works great when I use UDP or TCP transport, but when i switch to TLS it can't connect to the server. I think, this may be caused by self-signed server certificate.
So, is there a way to add my own certificates to Nokia Asha 501 ?

Hi.
I am trying to use TLS in voip environment and I am in a cross roads.
The guy who manages the asterisk server sent me two certificates: One CA.crt and another User.pem.
I have been able to make the asha 311 accept the CA.crt (after changing it to a ca.der file, according to this post: http://discussions.nokia.com/t5/Asha-and-other-Nokia-Series-30/HOW-TO-INSTALL-ROOT-Authority-CERTIFI...
But when I try to download the User.pem certificate, I get a message saying "no gateway reply".
So, I have two questions. ONe is, do I relly need these two certificates - is it that the asterisk guy is over complicting things?
The other is. if I need this pem file, how can I make the mobile accepct it?
thanks in advance.

Skype keeps giving me the same error over and over...

Good day everyone!
So I ran into this problem like a few months ago, and it is giving me the worse headache ever...
See... Everytime I log into Skype, it works fine for the first few times. After a couple of times that my skype has been restarted (When I go to work and have to shutdown my laptop) it logs in but doesn't send any messages and when I call someone it gets stuck (I can still press buttons but they don't give any actions) on the video call screen...
There is nothing I can do but kill the process, else this window of skype is in my screen the whole time.
When I (try) to login back into Skype, it says that my Skype has been already logged in...
I check my task manager and it says Skype.exe and when I try to kill the process it doesn't do anything...
This is really getting on my nerves...
What I do to fix this?
I press the Windows key and R for Run, in the box I type in %appdata% and delete the whole skype folder in there...
It works, for the first couple of starts...
After 1 restart or 2, it does the same thing again...
Please help me since this is really getting on my nerves...
With kind regards,
Delano
Solved!
Go to Solution.

techfreak wrote:
I've personally never run into this problem myself. The only time I've known it to occur on a continual basis is due to some 3rd party software. Sometimes triggered by normal operations, other times triggered by sleep modes.
The error occurs when Skype is restricted from updating it's logs and associated files when running. Something on your machine is more than likely interfering with Skype and disallowing it to write to your hard drive or temp files which it needs to do in order to function correctly and shutdown properly. This may be due to an AntiVirus/AntiMalware/Security software package or your computer's inability to awake properly from a sleep state. I would make sure that Skype is in the exception list of any type of Security software you are running. It would also be wise to limit the applications you automatically run on startup as one of them could be the culprit as well. You could also temporarily disable or modify your sleep settings on your computer to see if that's what triggers it.
It gives the same error in safe mode...
When I see the error, I am no longer able to write or delete any files in the %appdata%\Skype folder...
I've checked it and its because Skype is still running but doesn't respond to "Kill Task"

What T-Codes do Inbound 855 and Inbound 856 EDITransactions trigger?

Hi Folks,
I know inbound 856 is a call transaction method. It triggers VA01 transaction.
Likewise, Are inbound 855, 856 call transaction methods or not?
If yes, what transactions (T-Codes) do they trigger?
What are the process codes (inbound) for 855 and 856 inbound?
Thanks,
Creasy

Hi
for sending data from SAP to POS u can use WPMA tcode with the help of that Tcode you can send
Merchandise category information
Article master data
Control data
EAN
Bonus buys & coupons
Promotion discounts
Assortment lists
for inbound scenaro -
you can use standrad IDOC
like
WPUBON01 - Sales transactions, receipt-based
WPUUMS01 - Sales transactions, aggregated
WPUTAB01 - Means of payment (Upload day-end closing POS)
WPUKSR01 - Cashier statistics
WPUFIB01 - Financial transactions
WPUWBW01 - Goods movements
and process these IDOC with thep of Tcode BD20 or report RBDAPP01
PS: Reward Point
Thanks
Amit Shivhare

VoIP Monitoring and Management Tools

Does anyone have suggestion on VoIP monitoring and management tools for large enterprise voip infrastructure?
We are in the final phase of 8000 phone cut-over across 2 location (2clusters) and need some advice on choosing the right monitoring system.
I saw Qovia have Cisco approved monitoring tools.
Does anyone ever use this? any suggestions or advice?
Thanks in advance
Bonaldi Kresnanto

Bonaldi,
Monitoring voice quality is just one aspect of ongoing operations. It is equally (if not more) important to have the ability to remotely troubleshoot end-user problems, manage configuration changes, and perform active testing on a regular basis before users are impacted.
In addition, whenever you make changes to your IPT system (e.g. CallManager upgrades/patches, IOS upgrades, power outages) you'll definitely want the ability to verify the system is still operating as design. Using the active tests you can do this easily and in an automated fashion.
ClarusIPC Operations is the only product that addresses these needs. It is also just as valuable during the deployment as ongoing operations to help you make changes and deploy new phones with the same product you day-to-day.
I'd like to talk to you more about the product and how it can be used in your environment.
David Roberts
[email protected]

Create URL,Telephone number as paramater Account Search and result based on parameter of telephone number ?

Hi Experts,
create URL,Telephone number as paramater Account Search and result based on parameter of telephone number ?..
Previous my thread was locked...Now i want to say thanks to KALYANI L and Richa Dameja,..Now This code i have followed now its working fine..Thanks Great help to Kalyani L..and Richa Dameja.. i have implemented the DO_INIT_CONTEXT.....Now getting result...
As suggest Kalyani L i have implemented DO_INTI_CONTEXT method,Now its working fine....
Thanks For support.
Thanks
kalpana
Message was edited by: Andrei Vishnevsky
Disussion is locked.
Reason: Re: create URL,Telephone number as paramater Account Search and result based on parameter of telephone number ?

Hello Kalpana,
I've already locked your previous discussion and thought that I gave pretty clear warning.
First of all "do my job" posts are not welcomed on SCN.
Second point: you're incorrect in choosing SCN space with such questions. Here is a little which is related to IC in your task.
Third point is that if somebody has an answer to your exact question then he will give you it if he wants. There is no reason to post-post-post messages asking for help or hurry. Your "urgent requirement" is not the reason either.
Fourth one is: according to The SCN Rules of Engagement you need to do the search before posting. Almost all of your questions regarding this topic has an answer already.
Locking the discussion again. If you continue to post such questions I will need to report this situation to SAP CRM space editors and global moderators.

VoIP,data and video conferencing Design

hi
i want to design a network that support VoIP,data and Video conferencing for two offices (head offices and branch office ).What will be the best option ??
thanks
This topic first appeared in the Spiceworks Community

You can see this URL. The SRND is very useful. You can find information about the diferent models that you have to work. MCUs, GW, GK, endpoints and other components.
http://www.cisco.com/en/US/products/hw/video/ps1870/products_implementation_design_guide_book09186a00808f0e58.html

Skype won't let me click on user and not let me wr...

Skype won't let me click on user and will not let me write, I have the latest version and I have a pc withwindows 7, you have installed all updates, hear that the problem is with internet explorer, but how to fix it,because when I go to internet explorer and for example look for facebook and you feel the same, does notlet me log in nose

Try this:
Go to Windows Start -> All Programs -> Accessories -> Right click on Command Prompt, select 'Run as administrator'.
In the Command Prompt type: cd c:\windows\syswow64 and press Enter.
Next type this and then press Enter:
regsvr32 jscript.dll
Next type this and press Enter:
regsvr32 jscript9.dll
After each Enter you should receive a message saying that DllRegisterServer in XXX.dll succeeded.
Reboot your computer.
Make now the test showing if your Internet Explorer can execute JavaScript.
Open in your Internet Explorer this page and watch the response:
http://javatester.org/javascript.html

2 Office VOIP design, and PVDM usage

Dear colleagues,
i have 2 , A and B sites, with same spec's "number of users 16 with IP phones, and just VOIP needed , both will connected via Internet using a FW then IPsec to other branches",
for VOIP solution
A:will call PSTN via 4 analog lines, so i need 4 FXO
B:will call PSTN via VOIP provider so no FXO needed
each branch will call other branches using VOIP h323 trunks
my questions are,
1- PVDM needed?? for both sites?? why PVDM needed??
2-and if both sites will not call PSTN, directly, i.e if i used send landlines calls from the branch to my central office, i.e if i didnt use analog line nor VOIP provider to contact PSTN, IS PVDM NEEDED IN THIS CASE??
your help is appreciated
thanks

Hello Ahmed,
Hope I'm answering all your questions
A-hereunder example of VOIP senario and give me your commect.
1-if phone 1 call phone2, no transcoding needed?
No. Calls between phones controlled under the same CME will use G.711 and both phones supports G.711.
2-if phone 1 call phone 3, transcoding needed?, if needed, what CME is responsible for transcoding?, how it done, from both drection of voice traffic?
No. Call will be in G.729 and will work since Cisco IP phones (besides G.711 and other codecs) supports G.729. If for some reason phone 3 doesn't support G.729, then CME_b needs a transcoding resource to connect the call successfully.
B-Unity Express used only for Voice mail?, or it has another functions?
Supports Voicemail, Integrated Messaging, Auto Attendant, IVR and Fax.
C-if there is a CME router without PVDM module exist, can this router contain unity express?
Yes.
D-when i access unity express from CME CLI, i can use below command:
Yes. That command is used to access CUE via CLI, although you can use GUI as well.
E-what is this service-engine 0/0?, is it the AIM module??
Yes.
F-from the same CME router
1-there is PVDM2-32, and PVDM2-16, how i can choose the PVDM needed for my network?
Depending on the requirements (i.e. connections to PSTN, conferencing, transcoding, etc) you will need to calculate the total number of PVDMs. For this you can use the Cisco DSP Calculator: http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html
2-the AIM service engine 0, is it the Unity Express??
Yes.
3-when i access it , i use service engine 0/0, why 0/0, it is AIM 0 only, there is no 0/0?
The AIM is installed in slot 0, subslot 0.
F-what is related configuration to PVDM?, i.e, is there any provisioning configuration commands?
The most common is for conferencing and transcoding resources. Check this: http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/interop/intcnf2.html
G-when transcoding done by PVDM, it it need any configuration or done automatically?, i.e if G711 tranlated in real time to G729, is it use a specific commands in the run config, or just the existance of PVDM make the transcoding done automatically?
You configure the resources as shown in the link above. Once you get the resources registered and configured correctly, they will be invoked by CM/CME automatically when the situation demands.
H-is there is any config needed to provision AIM?, or just access the service engine and config the voice mail?
You first configure IP connectivity to the module. Check this: http://www.cisco.com/en/US/docs/voice_ip_comm/unity_exp/design/design21/cuenwinf.html#wp1008330
Next you can access it via CLI or GUI for configuration and integration to CM/CME.
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a008037f2a9.shtml#t11
http://www.cisco.com/en/US/products/sw/voicesw/ps5520/products_configuration_example09186a0080289ef0.shtml
Hope it helps. Thank you for the ratings.
-- Adrian.

Skype DECT telephone (e.g. Philips VOIP 855) and n...

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