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Changing from 3 digit extenstions to 4 on Cisco Unity Express

Posted on 2011-09-26
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Last Modified: 2012-05-12
We have created new 4 digit extensions for all our users and can dial them from within the organization.

The issue is I went into scripts and changed the extension length to 4 for the dial by extension, but when we dial in it clips the 4 digit extension to the first 3 digits entered.

I have looked through the running config of the system but have not found it defined anywhere at least that my novice eyes can see.
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Question by:LIKev
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5 Comments
 
LVL 15

Expert Comment

by:deepdraw
ID: 36600847
can you paste the dial peer config and maybe the rest of the phone stuff...

Greg
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Author Comment

by:LIKev
ID: 36601734
dial-peer voice 100 pots
 trunkgroup POTS
 preference 1
 destination-pattern 91[2-9]..[2-9]......
 forward-digits 11
 no register e164
 no sip-register
!
dial-peer voice 120 pots
 trunkgroup POTS
 preference 1
 destination-pattern [349]11
 forward-digits all
 no register e164
 no sip-register
!
dial-peer voice 121 pots
 trunkgroup POTS
 preference 1
 destination-pattern 9[349]11
 forward-digits 3
 no register e164
 no sip-register
!
dial-peer voice 130 pots
 trunkgroup POTS
 translation-profile outgoing OUTGOING-PSTN
 preference 1
!
dial-peer voice 1 voip
 translation-profile outgoing OUTGOING-SIP
 destination-pattern 91[2-9]..[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 20 voip
 translation-profile outgoing OUTGOING-SIP
 destination-pattern [349]11
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 21 voip
 translation-profile outgoing OUTGOING-SIP
 destination-pattern 9[349]11
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 30 voip
 translation-profile outgoing OUTGOING-SIP
 destination-pattern 9011.T
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 200 voip
 description CUE
 destination-pattern 6[^9].
 session protocol sipv2
 session target ipv4:*.*.*.*
 dtmf-relay sip-notify
 codec g711ulaw
!
dial-peer voice 202 voip
 translation-profile incoming INCOMING
 incoming called-number *******************
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 203 voip
 translation-profile incoming INCOMING
 incoming called-number **********...
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 2 voip
 translation-profile outgoing OUTGOING-SIP-800
 destination-pattern 91800[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 3 voip
 translation-profile outgoing OUTGOING-SIP-800
 destination-pattern 91866[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 4 voip
 translation-profile outgoing OUTGOING-SIP-800
 destination-pattern 91877[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 5 voip
 translation-profile outgoing OUTGOING-SIP-800
 destination-pattern 91888[2-9]......
 session protocol sipv2
 session target sip-server
 dtmf-relay rtp-nte
 codec g711ulaw
!
dial-peer voice 2000 voip
 destination-pattern 14..
 progress_ind setup enable 3
 progress_ind progress enable 8
 session target ipv4:10.202.1.1
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad
!
dial-peer voice 3000 voip
 session protocol sipv2
 session target sip-server
 incoming called-number .
 codec g711ulaw
 no vad
!
!
sip-ua
 no remote-party-id
 timers keepalive active 600
 registrar ipv4:*.*.*.* expires 3600
 sip-server ipv4:*.*.*.*
!
!
!
telephony-service
 sdspfarm conference mute-on 000 mute-off 111
 sdspfarm units 2
 sdspfarm transcode sessions 23
 sdspfarm tag 1 transcode01
 sdspfarm tag 2 conference01
 conference hardware
 authentication credential ***************************
 max-ephones 110
 max-dn 288
 ip source-address *.*.*.*  port 2000
 timeouts interdigit 6
 system message **************************
 url services http://*.*.*.* /voiceview/common/login.do
 url authentication http://*.*.*.* /CCMCIP/authenticate.asp  
 cnf-file location flash:
 load 7914 S00105000200.sbn
 load 7912 CP7912080003SCCP070409A
 load 7941 SCCP41.8-4-2S
 load 7942 SCCP42.8-4-2S
 load 7961 term61.default.loads
 load 7962 SCCP42.8-4-2S
 time-zone 12
 live-record 643
 voicemail 600
 max-conferences 8 gain -6
 moh flash:longwayhome.wav
 web admin system name ****************************************************/
 dn-webedit
 time-webedit
 transfer-system full-consult
 transfer-pattern .T
 secondary-dialtone 9
 after-hours block pattern 1 1900....... 7-24
 directory entry 1 600 name VoiceMail
 create cnf-files version-stamp 7960 May 10 2011 13:14:10
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LVL 15

Expert Comment

by:deepdraw
ID: 36602183
Im not sure what number you are trying to dial
please advise.

dial-peer voice 200 voip
 description CUE
 destination-pattern 6[^9].
 session protocol sipv2
 session target ipv4:*.*.*.*
 dtmf-relay sip-notify
 codec g711ulaw

if you are dialing 691 it would match the above..
so you would change to
 destination-pattern 6[^9]..

Greg
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Accepted Solution

by:
LIKev earned 0 total points
ID: 36709587
I solved it -- Recreated all the scripts and just set them all to allow 4 digit extensions. The orginal scripts were written with a Cisco script editor application so they could not be edited from the GUI express editor.  Once I wrote the new ones everything is correct.
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Author Closing Comment

by:LIKev
ID: 36898850
I am sure there is a way to change this setting in the original script but when looking at it in the cisco editor there does not seem to be an option.
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