Cisco Systems H.323/SIP Manual De Usuario

For Cisco Unified MeetingPlace Audio Servers systems that have both PSTN and IP interfaces, a dial 
plan should account for rollover from PSTN to IP ports and vice versa. For example, if you have a 
Cisco Unified MeetingPlace Audio Server system with 96 IP user licenses and 192 PSTN user licenses, 
the 97th caller to IP is automatically forwarded to a PSTN port by Cisco Unified CallManager through 
a voice gateway, rather than producing a fast busy signal. 

For additional information about mixed-mode configuration, see the Configuration Guide for 
Cisco Unified MeetingPlace Audio Server Release 5.3 at the following URL:

Use the following worksheet to create a a dial plan

.

From an IP phone to the IP-gateway server.

If the IP-gateway server is busy, Cisco Unified 
CallManager can forward calls to Cisco Unified 
MeetingPlace system PSTN through a voice gateway. 
You must configure Cisco Unified CallManager and 
the voice gateway to route this type of call. 

dial pattern____________________

A 4-digit number that does not conflict 
with a corporate phone extension number 
scheme.

From a PSTN phone to Cisco Unified MeetingPlace 
system PSTN.

If Cisco Unified MeetingPlace system PSTN is busy, 
the PBX or CO can forward calls to the IP-gateway 
server through Cisco Unified CallManager. You must 
configure the PBX or CO to route this type of call. 

dial pattern____________________

A 7- or 10-digit phone number that does 
not conflict with a corporate phone 
numbering scheme.

From Cisco Unified MeetingPlace system IP to an IP 
phone.

dial pattern____________________

Typically, the last four digits of the phone 
number. 

From Cisco Unified MeetingPlace system PSTN to a 
PSTN phone.

dial pattern____________________

Typically 9, if needed for an outside line, 
followed by either the 7- or 10-digit phone 
number.