TO-DO

test123

Barge and cBarge on CUCM

To add Barge or cBarge :

1. Barge: 

• Device / Phone : turn Privacy to OFF (default is ON)
• Device / Phone : set Built In Bridge to ON
• Device / Phone : select Single Button Barge to either Barge or cBarge (default is OFF)
• Device / Phone : select a Softkey Template. For Barge, a Standard User template is sufficient
• Device / Phone : add a shared-line DN, i.e   2001 for both test phones

The 2 test phones will now have a second shared-line button. On test-phone1, use shared-line to dial to another number.  Notice that test-phone2 second line will display a "double handset" sign on the shared-line and the shared-line button will turn RED.  Press the RED button on test-phone2 to barge in

1. cBarge: 

• Device / Phone : turn Privacy to OFF (default is ON)

• Device / Phone : set Built In Bridge to ON

• Device / Phone : select Single Button Barge to either Barge or cBarge (default is OFF)

• Device / Phone : select a Softkey Template. For cBarge, copy Standard User template to a new template and add cBarge softkey in call state "Remote In Use"

• Device / Phone : add a shared-line DN, i.e   2001 for both test phones

The 2 test phones will now have a second shared-line button. On test-phone1, use shared-line to dial to another number.  Notice that test-phone2 second line will display a "double handset" sign on the shared-line and the shared-line button will turn RED.  Press the RED button on test-phone2 to cbarge in. Note that cBarge use CFB resource , therefore MRG / MRGL must be set up and assigned to Device/Phone.

Tasks notes - important, go over

1. CUE and CME:

• add voice translation-rule/profile for VM numbers
• enable SIP - H323 interwork via  "voice service voip"
• bind SIP to interface gig0/0.230 (or same interface with CME)
• enable SIP-UA
• add Dial-Peer to make sure calls will reach CUE with SIP protocol and G711u - don't forget to include voice translation-profile
  
• add Dial-Peer to make sure MWI numbers will use G711u
• add num-exp (i.e num-exp 3500 21313500) to enable calls from PSTN to reach VM
• add voicemail number in CME and call-forward in ephone-dn
• add MWI on/off ephone-dn
• add Transcoder into CME if VM access from other sites (HQ or BR1) is needed
• use CUE Wizard for rest of task

1.  

AAR simplified in 5 minutes

In brief, AAR is for Automated Alternate Routing in case the WAN/IP network routing or bandwidth is restrictive to pass voice calls. Don't confuse this with SRST which is another subject

Procedure to setup AAR:

1. Decrease the bandwidth between sites so that a voice call wont have enough bandwidth, i.e change location bandwidth to 20K (minimum is 24K for g728 and 80K for g711).
2. on CUCM, Call-Routing / AAR-Group, add AAR-HQ, AAR-BR1 ...
3. Depend on the DialPlan configured previously, there 're 2 scenarios here for AAR routing. 
4. If you configured global Route-Pattern  \+!  then adding Translation-Pattern to route calls , there 're no need to add Dial_prefix between sites since the global RP  \+! should match all routes.
5. If you configured multiple RP for local/LD/INTL numbers, then calls using PSTN from HQ to BR1 should include prefix 9 (if BR1 External Phone Number Mask is 11 digits , i.e 1312301XXXX)  or prefix 91 (if BR1 External Phone Number Mask is 10 digits , i.e 312301XXXX)  and local calls inside HQ should use prefix 9 .
6. Update both HQ and BR1 Device/Phone AAR-CSS and AAR-Group with appropriate info. 
7. Update both HQ and BR1 Device/Phone/Line  AAR Settings with appropriate AAR Group.
8. Update both HQ and BR1 Gateway  AAR-CSS and AAR-Group with appropriate info.

Test by changing Location bandwidth of BR1 site to 20K. Calls from HQ phone to BR1 phone should be using PSTN with message "Network Congestion - Rerouting" appears on the HQ phone.