SIP - Setup - Skype

I just purchased a Nokia E66 and I am trying to do the SIP setup in order to use Skype. Does anyone know what to put as the Proxy Server and Realm? I am living in Qatar if that makes any difference.
Thanks for any and all help.

Skype doesn't use SIP. It uses its own protocol. You won't get Skype working by doing anything in your SIP settings.
There's probably an S60 application released by Skype that allows you to use this protocol on an E66. Alternatively, I know that fring allows you to use Skype protocol.
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Bbandtalk sip setup cant make outbound calls

hi i have extracted the sip settings for my bbandtalk service and loaded them into my 3cx server and it has registered fine. However, I cant make outbound calls. It comes up with a message saying 'sorry you cannot make or receive calls on this line'? any ideas? i have seen something about you cant use the bbandtalk sip setup unless you are on your bt broadband line....but if this was so, then how does the bbandtalk softphone client work?

Having similar problems ("Sorry, you cannot make or receive calls ..."). Using FreePBX and Asterisk and could use some help with the settings. Happy to take settings that go into other configuration files, but I'm currently stuck. Here's what I've got thus far (and it doesn't work -- tried several permutations, too):
type=peer
context=incoming_bbt
username=445603449781
authuser=445603449781
secret=B9R0WDKP9KAQMVX
host=62.239.169.148
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dtmfmode=rfc2833
Any help would be appreciated. Have other VOIP services working and BT seems to register fine.

Unable to perform call transfer or call park for an outbound call via SIP Trunk (SKYPE)

We have configured the SIP Trunk & SIP profile and successfull make outbound call through SIP Trunk (SKYPE). However, we are not able to perform call transfer or call park when the call is connected.
The scenario is:
A call to an phone number via SIP trunk, when call established, A perform call-transfer to B. After the call-transfer, the call Drop and Phone B show error code "Temp Fail"
When i select "enable MTP" in SIP trunk, we are able to call transfer and call park. But it limit the number of call session to 1.

You are probably running into some sort of Codec issue. IE, your phone is G.711 and the trunk is G.729. You will need to transcode the call at somepoint.

How can I setup skype on my iPhone to get notification alerts when someone contacts me via skype?

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Settings > Notification Center > (scroll down to Skype) Click on it > Then select what type of alerts you want your phone to give you. They are self explanatory.

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I can see Bonjour under bookmarks, but clicking on it doesn't do anything. I installed an update, so that's not the problem.
HELP!!!

My dvice flashed red for a few secongs, then started flashing green. It is still flashing green.
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External Calls , SIP Setup.

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SPA9000 / WIP330 SIP setup

Hi,
I have a SPA9000 at 10.10.1.5 on a network. Setting up SPA phones was not to complex.
My problem is with the WIP330. What should be the SIP Settings of the WIP330 in this setup? Is there any type of guide on setting up a WIP phone with the SPA9000 ?
Thank you all
Mario

Took me about a minute reading through the WIP330 manual to find it (makes one wonder doesn't it.. I have no such device myself).
On page 60 of the user guide you have the SIP settings.
proxy = ip-address-spa9k
proxy port: 6060
register proxy ip: ip-address-spa9k
register proxy port: 6060
outbound proxy ip:
outbound proxy port:
expire time: 300
phone number: 1234
username: 1234
password:
That should do it (when I didn't write anything.. leave it at its default). outbound proxy ip must not contain anything.. if it does, remove it.
If you can register but not make or take any calls, then set
outbound proxy ip: ip-address-spa9k
outbound proxy port: 6060
There are two essential pieces to tracking down a problem with your VoIP equipment:
The configuration of every device involved
SIP protocol traces
And don't forget: there's no such thing as giving too much information when describing a problem.

Unable to perform call transfer & call park through SIP Trunk (SKYPE)

The Scenario is:
I have set up a SIP trunk to SKYPE and we are able to make outbound call to a number via SIP Trunk.
After the call is established, when we tried to make call transfer, the call DROP and the phone at the other end shows error "Temp Fail".
I tried to "enable MTP" in SIP Trunk and We are able to perform call-transfer but it limits the call session to 1.
Anyone has facing the same issue?

MTP is needed to invoke supplementary functions like hold, transfer etc. Make sure that the MTP is checked on SIP trunk, MTP is assigned to the MRGL of the device pool on SIP trunk and has sufficient resources.
HTH
Manish

Nokai E75 VOIP/SIP Setup for WLAN: sipgate.de - ...

Had problems setting up the VOIP Client on my E75 for the SIP Provider sipgate.de: Test calls to their 10005 or 10000 number worked without problems, but for outgoing calls to normal landline numbers no audio stream was connected through (ringback was ok):
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   - WLAN (linksys or AVM-Fritzbox)
   - Provider Tcom/Avego or 1&1
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   - NAT FIREWALL SETTINGS -> DOMAIN PARAMETERS
       -> select sipgate.de
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if you don't have the "Advanced VOIP Settings" Menu, you need to load a special sip addon application. Documentation and application is available at
    forum.nokia.com
Search for "SIP VOIP SETTINGS"
If the above tweak also works for other WLAN Router or even UMTS Settings, I don't know.
With regards to sipgate.de --> havn't used it yet a lot, but the good thing is it's not a standalone client, but as it's reusing the Nokia VOIP Client it nicely integrates into the phone (same contact list, button to choose internet vs. mobile dialing, ..
hope the above saves some people some time
jo

Native VoIP/SIP is the main reason I was so pleased to switch to the E72 (I use voiptalk.org myself).
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Macbook Pro Mikrophone Echo for SIP and Skype

Hi All
I have a Macbookpro retina 2013. When I try to Skype or use my SIP phone apple phone app, The receiver of my call has an Echo. Should I take the laptop back?

First your machine is NOT affected by a virus because none as in zero, nada, zip, nope exist for OS X. What it sounds like to me though is 1 of 2 possibilities, both related to your HD. First is it's too full, Mac's need a minimum of 10-15% of the disk available at all times. For example if your drive is 250GB it always should have at least 25-38GB available. If you aren't sure how much yours has in Finder click it and then type Command + I and it will tell you the size and how much is available.
The next thing I think it might be is that it's failing. I would recommend running Apple Hardware Test in Extended Mode immediately and see if it reports any errors.
Roger

SIP setup and the 'sip phone' symbol on E71

Hey....
OK SIP users..this question is for you.
Iam running my E71 and have activated and paid for a SIP account (voiptalk) and the setup went well last week, and I have made a few successful SIP calls from the E71.... good stuff !
I am calling landline numbers in the UK using SIP (VoipTalk) through FRING... no problems...
My question is... the little SIP-Phone symbol that was present on the top left of the screen has now disappeared... but I can make SIP calls...
Erm... why has the symbol gone ?
What does that symbol mean in reality, as I can continue to make SIP calls...
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Harry

Okay thats a question more to the phone's behavior rather then SIP settings.
Looks like your phone keep dropping out symbols.
Have you tried to reboot your phone lately by swithing it off removing the battery and put it back together after a couple of minutes.
Sometimes that solves a lot of issues.
Try it and see if it brings back the symbol.
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SIP setup with Media5-fone and CUCM7

I think I have configured it all correctly and I have viewed the netcraftsmen website but it differs a little.
When I run a sip trace from Media-fone5 i see an error : Warning: 399 (CUCM name) " Line not Configured"
I can not find out what line needs to be configured.. Any ideas?

Gene,
Actually, I just loaded Media5 on my iPod Touch for testing so we can add it to our blog. I haven't finished testing, but I am able to get the phone to register. A full blog is forthcoming but you may want to try this in the interim. Note, we are assuming the same basic approach as you see for the various Droid and iPhone clients Hailey and I write about in the blog.
Under the account settings main page:
Username: set this to the line DN you have assigned to the SIP device in CUCM (e.g. 7035551234)
Password: set this to the digest you assigned to the end user you created in CUCM (e.g. C1$co12345)
Servers config page:
Put the IP address of the server you want to register to. Keep the port info the same.
Advanced config page:
Auth Name: set this to the user ID assigned to the end user you created in CUCM (e.g. mysipuser)
DTMF Method: You should be fine with RFC 2833 (default) but I have not tested
Again, I just loaded this yesterday and only tested registration. I plan on going back to the config once I get some free cycles. So, we will have a write up on the blog sometime in the next week.
Let us know if the cursory overview above gets you pass your current issue.
HTH.
Regards,
Bill

Not able to setup Skype To Go numbers (404 error)

Hi,
I am getting page not found when I try to register a India number to a Singapore mobile number. The error is occurring in the last page (Make a call ). Can anyone help on this.
http://www.skype.com/error/404.html

OS ?
Does your IP address declared in /etc/hosts (on Unix/Linux) or \WINDOWS\system32\drivers\etc\hosts (for Windows) ?
Nicolas.

Skype for Business - Is it Ready for Business?

Hello Support,
We are evaluating Skype as a potential main solution for all our telephony systems.
We currently see many issues in the Gui Client and whole management proposal from our perspective.
Not sure how others have dealt with it.
With all respect, to us it seems the Business version of Skype is really limited for Companies with few employees.
Some of the problems we see.
1) Can't manage centrally the Phone/Contacts directory.
This means that adding an Employee account in the Company forces every single Employee having a Skype account and add that user in their own "Directory". This is something that is natural to be done centrally and pushed to the end-users.
2) The number of restrictions on the Client are definitely not sufficient in a Corporate Environment (end-users should not be able to: Add, Change, Remove or Block Contacts, modify any of the settings or views of the GUI etc). Basically the management console should control every aspect of the managed clients GUI and functionality. Best if those settings could be "pushed" to the end-points instead of dealing with GPO as it is more flexible. You may have totally separate domains (not trusting each other) and GPOs might not be the best way to distribute client settings. GPO would not work on Corporate Phones for example.
3) The system should have the ability to prevent a user to log off from Skype and log on with their personal user. In a Centrally Managed environment you don't want people to mess with the setup. Also, user name and password of their "Skype Client" may not be shared. There are many businesses in the Energy Sector where machines are setup x role and not x user (so when one Employee leaves the account will not be removed).
4) The msi should have the ability to embed (or pass via parameter) the main account where the Client is registered. In the Skype Manager I should then be able to see how many "devices" have been currently installed for the same user and manage/approve them accordingly.
The management office may not approve the Company Skype account on a device that does not belong to "Corporate".
5) Calls may be done world-wide but Support isn't.
Some countries (Singapore for example) are simply refused a chat session with the Skype Support with a "Not supported in your own Country" and redirected to a "Self Service" FAQ.
6) Skype conflicting with the Lync proposition and may be completely replaced by Lync in the "near?" future.
Not clear if MS wants to make money on the "voice" platform, on licencing or both.
Skype would be an attractive platform if those problems could be resolved but if potential business clients are pushed to Lync then the competition becomes Cisco Unified Communication, Huawei E-Space, Polycom UC etc (totally different ballgame). A company going through that complex setup would then also have to consider free open source such as Asterisk (using any free XMPP client and SIP integration).
So, if one wants to go completely "free" and have a very simple setup Skype could be an excellent choice, but the current issues (from our perspective) hinder the chances a mid-size / large enterprise adoption of the platform.
I have looked at the SkypeKit solution that may be used to completely re-create the GUI and probably eliminate many of the previous listed points, but rewriting the GUI is not a small task and you end up getting stuck with a proprietary "in-house" solution.
It'd be nice to see Microsoft views in this matters.
Thank you.

Hello,
I am an export manager consultant for a French company. Starting in September, I shall be in charge of sales personnel in France as well as international sales primarily those from North America. I want to set up bi-monthly video chats between the sales team, the office and be able to talk with all people (maximum 8 people) when I am travelling, am in the office, or from abroad. Unfortunately, I need to be reacheable 24/7. I also have my company.
The office has PCs, two IPads and I have a MacBook Air.
The Company has 4 personal accounts. I have my own personal account at the moment.
Should we all merge these accounts under one roof? Can I add my personal account to this eventual business account?
What is the best solution to be able to video chat, have a minimum credit on each salesperson's iphone and be able to manage all this without too much trouble? (I have not been able to find the requisites needed for a Skype business account and its advantages over a personal account.
Thanks for your replies.
Jacques

Skype connect, outbound failing

I recently setup skype connect for product testing on an NEC SV8100.
I can recieve call ok.
When i dial out the other end rings, but no RTP audio, then it gives engaged.. I still get charged for each call !!
Done a wireshark trace :
No.     Time        Source                Destination           Protocol Length Info
      1 0.000000    172.10.0.1            63.209.144.201        SIP/SDP 934    Request: INVITE sip:[email protected], with session description
Frame 1: 934 bytes on wire (7472 bits), 934 bytes captured (7472 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      2 0.261931    63.209.144.201        172.10.0.1            SIP      521    Status: 407 Proxy Authentication Required
Frame 2: 521 bytes on wire (4168 bits), 521 bytes captured (4168 bits)
Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      3 0.328555    172.10.0.1            63.209.144.201        SIP      455    Request: ACK sip:[email protected]
Frame 3: 455 bytes on wire (3640 bits), 455 bytes captured (3640 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      4 0.457122    172.10.0.1            63.209.144.201        SIP/SDP 1175   Request: INVITE sip:[email protected], with session description
Frame 4: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      5 0.999458    172.10.0.1            63.209.144.201        SIP/SDP 1175   Request: INVITE sip:[email protected], with session description
Frame 5: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      6 2.017736    172.10.0.1            63.209.144.201        SIP/SDP 1175   Request: INVITE sip:[email protected], with session description
Frame 6: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      7 4.000270    172.10.0.1            63.209.144.201        SIP/SDP 1175   Request: INVITE sip:[email protected], with session description
Frame 7: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      8 8.019283    172.10.0.1            63.209.144.201        SIP/SDP 1175   Request: INVITE sip:[email protected], with session description
Frame 8: 1175 bytes on wire (9400 bits), 1175 bytes captured (9400 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060)
Session Initiation Protocol
No.     Time        Source                Destination           Protocol Length Info
      9 11.947857   63.209.144.201        172.10.0.1            UDP      214    Source port: 26998 Destination port: 10020
Frame 9: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
User Datagram Protocol, Src Port: 26998 (2699, Dst Port: 10020 (10020)
Data (172 bytes)
No.     Time        Source                Destination           Protocol Length Info
     10 11.948226   172.10.0.1            63.209.144.201        ICMP     70     Destination unreachable (Port unreachable)
Frame 10: 70 bytes on wire (560 bits), 70 bytes captured (560 bits)
Ethernet II, Src: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff), Dst: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f)
Internet Protocol Version 4, Src: 172.10.0.1 (172.10.0.1), Dst: 63.209.144.201 (63.209.144.201)
Internet Control Message Protocol
No.     Time        Source                Destination           Protocol Length Info
     11 11.967964   63.209.144.201        172.10.0.1            UDP      214    Source port: 26998 Destination port: 10020
Frame 11: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits)
Ethernet II, Src: Cisco-Li_cd:0b:1f (98:fc:11:cd:0b:1f), Dst: NecInfro_0f:4f:ff (00:60:b9:0f:4f:ff)
Internet Protocol Version 4, Src: 63.209.144.201 (63.209.144.201), Dst: 172.10.0.1 (172.10.0.1)
User Datagram Protocol, Src Port: 26998 (2699, Dst Port: 10020 (10020)
Data (172 bytes)

Try to reset all Skype settings.
Quit Skype or use Windows Task Manager to kill any Skype.exe process. Go to Windows Start and in the Search/Run box type %appdata% and then press Enter or click the OK button. The Windows File Explorer will pop up. There locate a folder named “Skype”. Rename this folder to something different, e.g. Skype_old.
If you are on the latest Skype 6.5/6.6 version, then do also this:
Go to Windows Start and in the Search/Run box type %temp%\skype and then press Enter or click the OK button. Delete the DbTemp folder.
Restart Skype.
N.B. If needed, you will still be able to re-establish your call and chat history. All data is still saved in the Skype_old folder.

SIP - Setup - Skype

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