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Skype connect with Cisco Cube Config Template
Skype connect with Cisco Cube
ip domain name corp.XXXX.org
ip name-server 8.8.8.8
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol none
no fax-relay sg3-to-g3
h323
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
registrar server
redirect contact order best-match
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g729r8
!
dial-peer voice 1014 voip
service session
destination-pattern 00T
progress_ind progress enable 8
modem passthrough nse codec g711ulaw
session protocol sipv2
session target dns:sip.skype.com
voice-class codec 1
dtmf-relay rtp-nte
no vad
!
sip-ua
credentials username XXXXXXXX password 7 XXXXXXXX realm sip.skype.com
authentication username XXXXXXXX password 7 XXXXXXXX
registrar dns:sip.skype.com expires 60
sip-server dns:sip.skype.com
CUCM
====
Under Route Pattern > Calling Party Transformations > Calling Party Transform Mask
Enter the username
Some “show” cmd
Router1#sh sip-ua service
SIP Service is up
!
Router1#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
.* 10 64 no
XXXXXXXXXXXXXX -1 19 yes
!
Router1#
Router1#
Router1#show sip-ua register status
Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
.* 10 64 no
XXXXXXXXXXXXXX -1 19 yes
Router1#
Router1#
Router1#show sip-ua registration status
SIP User Agent Status
SIP User Agent for UDP : ENABLED
SIP User Agent for TCP : ENABLED
SIP User Agent for TLS over TCP : ENABLED
SIP User Agent bind status(signaling): ENABLED 10.x.x.x
SIP User Agent bind status(media): ENABLED 10.x.x.x
SIP early-media for 180 responses with SDP: ENABLED
SIP max-forwards : 70
SIP DNS SRV version: 2 (rfc 2782)
NAT Settings for the SIP-UA
Role in SDP: NONE
Check media source packets: DISABLED
Maximum duration for a telephone-event in NOTIFYs: 2000 ms
SIP support for ISDN SUSPEND/RESUME: ENABLED
Redirection (3xx) message handling: ENABLED
Reason Header will override Response/Request Codes: DISABLED
Out-of-dialog Refer: DISABLED
Presence support is DISABLED
protocol mode is ipv4
SDP application configuration:
Version line (v=) required
Owner line (o=) required
Timespec line (t=) required
Media supported: audio video image
Network types supported: IN
Address types supported: IP4 IP6
Transport types supported: RTP/AVP udptl
!
!
Router1#
Router1#
Router1#show sip-ua registration connections tcp brief
Total active connections : 1
No. of send failures : 0
No. of remote closures : 2
No. of conn. failures : 0
No. of inactive conn. ageouts : 301
Max. tcp send msg queue size of 1, recorded for 10.x.x.x:55020
-------------- SIP Transport Layer Listen Sockets ---------------
Conn-Id Local-Address
=========== =============================
2 [10.x.x.x]:5060
Router1#
Router1#
Router1#show sip-ua registration connections udp brief
Total active connections : 2
No. of send failures : 0
No. of remote closures : 0
No. of conn. failures : 0
No. of inactive conn. ageouts : 277
-------------- SIP Transport Layer Listen Sockets ---------------
Conn-Id Local-Address
=========== =============================
2 [10.x.x.x]:5060
Can you please share your redacted full debug? I’m troubleshooting why my 2901 w/ CME is not passing the header correctly. I believe it is because I use an “8” to dial out.
Why we need to add SIP trunk on CUCM. Correct me if I am wrong, we just need to create trunk from CUCM to GW with respective rp,rl etc, and then vgw iwill manage call routing itself