Skype now offering SIP?

[leejor]

I had an email from Skype the other day and it looks like they are, or will soon be, offering SIP, from what I gather, for businesses only.
Didn't really delve into it but it seems that Skype is looking to expand their customer base.

Came across the link to it...http://www.skype.com/intl/en/business/products/pbx-systems/sip/get-it-now/#paddedContent

 

[igor.snezhko]

Yes, for money. But 3CX Gateway is free. :lol:

 

[mike.petersen]

Has anyone tried this out? I'm currently working on setting up incoming/outgoing calls with the 3cx Skype Gateway, but am thinking this might be a better option.

 

[MP_Swede]

Hi,

I have Skype for SIP now, incoming calls both landline and from other Skype clients works very well. (Even if it takes 30 sec before it comes to the PBX from Skype clients)

Outgoing calls I'm still struggling with as I get below error

12:10:22.634 [CM503008]: Call(28): Call is terminated
12:10:22.580 [CM503020]: Normal call termination. Reason: Not found
12:10:22.580 [CM503016]: Call(28): Attempt to reach <sip:[email protected]> failed. Reason: Not Found
12:10:22.580 [CM503003]: Call(28): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:193.120.218.68:5060
12:10:22.099 [CM503025]: Call(28): Calling VoIPline:0708215965@(Ln.10001@Skype)@[Dev:sip:[email protected]:5060]
12:10:21.946 [CM503004]: Call(28): Route 1: VoIPline:0708215965@(Ln.10001@Skype)@[Dev:sip:[email protected]:5060]
12:10:21.942 [CM503010]: Making route(s) to <sip:[email protected]>
12:10:21.940 [CM505001]: Ext.500: Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [DIP Phone 450010630000000] PBX contact: [sip:[email protected]:5060]
12:10:21.937 [CM503001]: Call(28): Incoming call from Ext.500 to <sip:[email protected]>

 

[MP_Swede]

I just notised the line
12:10:22.580 [CM503003]: Call(28): Call to sip:[email protected]:5060 has failed; Cause: 404 Not Found; from IP:193.120.218.68:5060

Why does it say "Call to sip:*" ?

On my working SIP trunks I do not get this extra line and they work perfect.

 

[leejor]

Could you be sending a number format that they don't accept or just don't understand? The "0" prefix might be a standard within the PSTN network in your country but for Skype you may have to add a few digits to the beginning of your outgoing number.
http://www.tekelec.com/SIPReferenceGuide/mapping_ISUP_cause_codes_to_SIP-I_SIP-T_responses.asp