SIP for Skype Beta - configuration instructions

[robpemberton]

Has anyone been able to get the SIP for Skyp Beta working? Are there instructions available for setting up this as a voip provider?

 

[mfm]

Hi,

Have a look this article, it should start you into the right direction:
http://www.3cx.com/blog/news/3cx-gateway-for-skype/

 

[robpemberton]

Thanks for the reply -

Thats' great info for setting up the 3cx gateway to skype.

I'm looking to configure - Sip to Skype

Is there documentation on that?

 

[leejor]

If it is still a Beta then I would assume that they aren't releasing any of the needed info (server name, etc) to the general public just yet. If you have signed up with them then you would probably be given the standard information that you would need as with any SIP provider. That could then be "plugged into" the generic SIP trunk setup. Perhaps someone on the board is using Skype (the SIP version) and can fill us in?

 

[robpemberton]

Here is the guide from skype -
http://download.skype.com/share/business/guides/skype-for-sip-quick-start-guide.pdf

I have used the generic provider to try to set this up and it isn't working.

Anyone have any ideas of what to try?

 

[fceledon]

Any news about this? I cant' make it work.... (sip.xx.com is my server)
The registration is OK

11:32:53.915 [CM503020]: Normal call termination. Reason: Not found
11:32:53.915 [CM503016]: Call(37): Attempt to reach <sip:[email protected]> failed. Reason: Not Found
11:32:53.905 [CM503003]: Call(37): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:63.209.144.201:5060
11:32:52.763 [CM503025]: Call(37): Calling VoIPline:0115625551212@(Ln.10001@skype)@[Dev:sip:[email protected]:5060;rinstance=dfcd15b3f45597bb]
11:32:52.453 [CM503004]: Call(37): Route 1: VoIPline:0115625551212@(Ln.10001@skype)@[Dev:sip:[email protected]:5060;rinstance=dfcd15b3f45597bb]
11:32:52.382 [CM503010]: Making route(s) to <sip:[email protected]>
11:32:52.312 [CM505001]: Ext.110: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sipura/SPA941-4.1.8] PBX contact: [sip:[email protected]:5060]
11:32:52.162 [CM503001]: Call(37): Incoming call from Ext.110 to <sip:[email protected]>

 

[SY]

Any news about this? I cant' make it work.... (sip.xx.com is my server)
The registration is OK

11:32:53.915 [CM503020]: Normal call termination. Reason: Not found
11:32:53.915 [CM503016]: Call(37): Attempt to reach <sip:[email protected]> failed. Reason: Not Found
11:32:53.905 [CM503003]: Call(37): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:63.209.144.201:5060
11:32:52.763 [CM503025]: Call(37): Calling VoIPline:0115625551212@(Ln.10001@skype)@[Dev:sip:[email protected]:5060;rinstance=dfcd15b3f45597bb]
11:32:52.453 [CM503004]: Call(37): Route 1: VoIPline:0115625551212@(Ln.10001@skype)@[Dev:sip:[email protected]:5060;rinstance=dfcd15b3f45597bb]
11:32:52.382 [CM503010]: Making route(s) to <sip:[email protected]>
11:32:52.312 [CM505001]: Ext.110: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sipura/SPA941-4.1.8] PBX contact: [sip:[email protected]:5060]
11:32:52.162 [CM503001]: Call(37): Incoming call from Ext.110 to <sip:[email protected]>

Selected log message provides information for you.
You can ask your provider about how to correctly form a dialed number to be able to reach the destination.

Thanks

 

[fceledon]

Thanks, indeed the solution was pretty simple:
On the skype sip rule, if the number starts with "011" remove 3 and prepend a "+" sign.
.... now working fine!!