Call Manager & Skype connect via CUBE

In the previous topic I described how to make calls from CCM to Skype Connect. What was rightly stated that for such purposes you need to use Cisco CUBE . Now it's time and memory to turn the old 2811 into a cube and make kosher integration.

Suppose that the money on Skype is already there. SIP account is created and we know the login and password for SIP.

Settings at 2811.

voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
allow-connections sip to sip
sip
registrar server
')
ip address trusted list - needed for Toll-Fraud Prevention . Otherwise there will be a 403 error for all calls.

If the trunk should be created from a specific interface, then you need to register
voice service voip
sip
bind control source-interface bla-bla-bla
bind media source interface bla-bla-bla

SIP settings. You can copy by replacing 99XXXXXXXXXXXXX and 7 YYY with your SIP username and password.
sip-ua
credentials username 99XXXXXXXXXXXXX password 7 YYY realm sip.skype.com
keepalive target dns: sip.skype.com
authentication username 99XXXXXXXXXXXXX password 7 YYY realm sip.skype.com
no remote-party-id
max-forwards 10
retry invite 1
retry response 1
retry bye 1
retry cancel 1
retry register 10
timers trying 250
timers expires 60000
timers connect 250
timers disconnect 250
mwi-server dns: sip.skype.com expires 3600 port 5060 transport udp unsolicited
registrar dns: sip.skype.com expires 3600
sip-server dns: sip.skype.com
no suspend-resume
connection-reuse
host-registrar
handle-replaces

Check registration
show sip-ua register status
Line peer expires (sec) registered P-Associ-URI
========================================= ======== ===
99XXXXXXXXXXXX -1 50 yes

If not, we smoke debug ccsip messages . I got it right away. There may be problems with firewall and NAT. On the firewall, I only have the usual dynamic nat, nor any static. Well, UDP for CUBE above port 1024 is allowed.

voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8

Without "voice class codec" Media Type (s) Unavailable and Resource unavailable, unspecified errors were issued.

dial-peer voice 10 voip
session protocol sipv2
session target dns: sip.skype.com
incoming called-number 1 ...
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 20 voip
destination-pattern 1 ...
session protocol sipv2
session target ipv4: ccm
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 30 voip
destination-pattern .T
session protocol sipv2
session target dns: sip.skype.com
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 40 voip
session protocol sipv2
session target ipv4: ccm
incoming called-number .T
voice-class codec 1
dtmf-relay rtp-nte
no vad

dial-peer 10 - Receiving from skype. Three-digit number with 1
dial-peer 20 - Peer on ccm.
dial-peer 30 - Peer on skype for outgoing calls
dial-peer 40 - Receiving from ccm

SIP Trunk on CCM by default with the exception of Caller ID DN, where you need to put the login from the SIP Profile.

in Skype Manager we add skype account, we register for it internal number and we try to call. I got it right away.
Well and in CCM we do Route Patterns on our trunk.
From the short test, it turned out that the calls from skype do not reach everyone: skype is buzzing, but there is nothing on cisco. Call quality at 4.

In general, will live for as long as calls from skype