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2901 CME-FWD of 3 FXO lines to cell-enable/disable with key on phone

I have a very simple CME site running on a stand alone 2901 router with 6-7821 SIP phones and 3 FXO lines, no Auto-Attendant.  The 3 lines hunt inbound by the carrier and ring a hunt group, that all 6 extensions are a member of, so that anyone can answer an incoming call.  The manager at the site would like to be able to forward the 3 FXO incoming lines out to his cell during off hours or heavy storms when the office is closed.  He also would like a key on his 7821 SIP phone to enable and disable this functionality.  The router is running IOS 15.5(3)M1 with CME v10.5.

Is this functionality possible on a CME without reconfiguring the entire system?

Or would simply forwarding his 7821 extension essentially do the same thing since all 3 lines ring his extension (as well as the other 5)?

Brandon Buffin

One option is to use single number reach. This would ring an external phone simultaneously with the internal extensions. The user would have a mobility softkey to turn the feature on/off.


SNR is not supported over FXO lines officially, but if that works for you it is an option.

Alternatively, your proposed solution would work if you simply configure call forward all on the phone, assuming you still have enough outbound lines to send the call out and that you are OK with presenting the caller ID of the individual FXO line to the called party as that would be the only option with FXO trunks.


I am trying to set up forwarding of the users phone to external.  Since all incoming lines ring his prime extension, this should work.  They are definitely limited by the 3 FXO lines and understand the potential trunk resource issue this may create.

The user is getting a fast busy when he uses the "forward all" on his phone and enters his external cell as the destination.  He is entering it exactly as he is able to dial his cell with a leading "91".  I am obviously missing something in the config to allow this forward to external functionality.

My apologies if this is a simple fix, new to Cisco.  Below is a brief config.  Where there are X's the numbers are just masked.  We are using 10 digit extensions (long story).  Where it says NAME and PASSCODE, these are also masked.  Thanks.

voice register dn 1
call-forward b2bua busy 1000
call-forward b2bua noan 1000 timeout 17
name Office 1
voice register pool 1
busy-trigger-per-button 2
type 7821
number 1 dn 1
dtmf-relay rtp-nte
codec g711ulaw

voice-port 0/1/0
trunk-group outbound
connection plar 2000
voice-port 0/1/1
trunk-group outbound
connection plar 2000
voice-port 0/1/2
trunk-group outbound
connection plar 2000

dial-peer voice 1 pots
description XXX-XXX-XXXX
port 0/1/0
no sip-register
dial-peer voice 2 pots
description XXX-XXX-XXXX
port 0/1/1
no sip-register
dial-peer voice 3 pots
description XXX-XXX-XXXX
port 0/1/2
no sip-register
dial-peer voice 10 pots
trunkgroup outbound
description 10 Digit Local Out
destination-pattern 9[2-9].........
forward-digits 10
dial-peer voice 11 pots
trunkgroup outbound
description Long Distance Out
destination-pattern 91[2-9].........
forward-digits 11
dial-peer voice 110 pots
trunkgroup outbound
description International Out
destination-pattern 9011T
prefix 011
dial-peer voice 911 pots
trunkgroup outbound
destination-pattern 911
forward-digits all
dial-peer voice 9911 pots
trunkgroup outbound
description Emergency Call
destination-pattern 9911
forward-digits 3
dial-peer voice 7 pots
trunkgroup outbound
description 7 Digit Local Out
destination-pattern 9[2-9]......$
forward-digits 7
dial-peer voice 500 voip
description Unity Express-Voicemail
destination-pattern 1000
session protocol sipv2
session target ipv4:X.X.X.X
incoming called-number .
dtmf-relay sip-notify
codec g711ulaw
no vad

voice service voip
ip address trusted list
no ip address trusted authenticate
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0
registrar server expires max 1200 min 300

authentication credential NAME PASSWORD
url services http://X.X.X.X/voiceview/common/
url authentication http://X.X.X.X/CCMCIP/authenticate.asp
time-zone 3
voicemail 1000
max-conferences 8 gain -6
web admin system name NAME password PASSWORD
transfer-system full-consult

I just found this in the CME admin guide so I may not be able to use the forwarding:

• Call forwarding is invoked only if that phone is dialed directly. Call forwarding is not invoked when
the phone number is called through a sequential, longest-idle, or peer hunt group.
• If call forwarding is configured for hunt group member, call forward is ignored by the hunt group.
• Calls from an internal extension to an extension which is busy, is forwarded to the SNR destination
even if no forward local-calls is configured under the Directory Number.

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