[cisco-voip] Why is my CUBE sending my dial-peers instead of login information to Skype Connect?

[Stephen Welsh]

I've configured Skype Connect using a CUBE config (SIP2SIP) directly on the internet, I did it for the same reasons are yourself, however I was disappointed at the cost (to actually place calls) and the fact you cannot call to a Skype User ID from the IP-PBX (you can do the reverse by calling your Skype Connect ID number from any Skype Client). I was hoping to do some Single Number Reach with my Skype ID :(

Below are the relevant parts from my working config, however you need to add your own security statements (I'm not responsible for your phone bill :), you will most likely be exposing port 5060/61 to the Wide Wicked Web, and I've seen a LOT of registration attempts hit my router....

Also,
I still haven't managed to get DTMF working for outbound calls (dam no conf calls via Skype Connect), but it does work inbound, I traced a thread that suggested the 'voice-class sip dtmf-relay force rtp-nte' command, but it didn't work for me, but I may not have applied it correctly. If you do figure outbound DTMF, let me/us know ;)

Note: You can route/match on the Skype In PSTN number, it will be presented as E.164 without the '+'
Note: You can also route Skype connect ID 99........ etc. 
Note: Your CLI needs to be your Skype Connect ID, so make sure to set it on the outbound Route Patterns in UCM etc. otherwise Skype will reject it (needs to be in the From header in the SIP INVITE)
Note: The outbound destination number sent to Skype can be E.164 (without the '+'), or E.164 with 00 instead of the '+'

Thanks

Stephen


voice service voip 
 allow-connections sip to sip
 redirect ip2ip
!
voice class codec 1
 codec preference 2 g711ulaw
 codec preference 3 g729r8
!
ip nat sip-sbc

!         
dial-peer voice 1 voip
 destination-pattern <Skype In Number PSTN i.e. 12121234567>
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target ipv4:<UCM IP>
 dtmf-relay rtp-nte
!
dial-peer voice 2 voip
 destination-pattern <Skype Connect ID i.e. 99…….>
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target ipv4:<UCM IP>
 dtmf-relay rtp-nte
!
dial-peer voice 100 voip
 description SIP Trunk to Skype
 destination-pattern 00T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
dial-peer voice 101 voip
 description SIP Trunk to Skype
 destination-pattern +T
 voice-class codec 1
 voice-class sip dtmf-relay force rtp-nte
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 no vad
!
sip-ua 
 credentials username <Skype Connect ID i.e. 99…….> password 0 <Skype Connect Password> realm sip.skype.com
 authentication username <Skype Connect ID i.e. 99…….> password 0 <Skype Connect Password>
 nat symmetric role passive
 nat symmetric check-media-src
 registrar dns:sip.skype.com expires 65535
 sip-server dns:sip.skype.com


On 1 Mar 2011, at 19:25, Robert Kulagowski wrote:

> I'm messing around with Skype Connect, because it seems like a cheap /
> easy (hah!) way to get some experience without paying for multi-year
> contract from a vendor.
> 
> However, my router appears to be sending my dial-peers during the
> registration rather than the username that is in my Skype Manager
> account!
> 
> Does anyone have a sanitized CUBE configuration that works with Skype
> Connect?  My CUCM is talking to CUBE via H.323, and that part appears
> to be OK.
> 
> Skype tech support chat told me to also try IP authentication rather
> than SIP authentication, but that's not working either.
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