02-13-2009 04:26 AM - edited 03-15-2019 04:12 PM
Hi
I am working with a customer who has a UC500 connected to a SIP trunk in the UK. We are having a problem with the SIP provider because the removing the leading zero on the incoming number so when the call come through it is displayed without the first digit on the IP phone display. The problem for the user is they cannot press redial as the outgoing number is then incorrect. Is there a way of prefixing an inbound number and adding the missing digit?
Thanks
Ian
Solved! Go to Solution.
02-16-2009 04:27 AM
Thank you, I've rated your post as you took further time on an already resolved issue.
02-13-2009 04:39 AM
voice translation-rule 10
rule 1 // /0/
voice translation-profile add0
translate calling 10
dial-peer voice XX voip
incoming called-number ZZZ
translation-profile incoming add0
02-13-2009 04:49 AM
I will try this, thank you very much for being so prompt
02-13-2009 04:50 AM
You are welcome, remember to rate useful posts with the scrollbox below.
02-13-2009 05:52 AM
Hi
This doesn't seem to be working. I have configured it as you suggested but am still dropping the zero. Can I post you my config?
02-13-2009 05:57 AM
Possibly it is not matching the DP.
Do you have a single voip DP with incoming called-number ?
02-13-2009 06:12 AM
Hi
Not sure what you mean by DP. We have a series of DDI numbers being delivered over a SIP trunk with the following config
voice translation-rule 4
rule 1 /441612413849/ /3849/
!
voice translation-rule 5
rule 1 /441612413850/ /3850/
!
voice translation-rule 23
rule 1 /441612417746/ /7746/
rule 2 /441612417747/ /7747/
rule 3 /441612417748/ /7748/
!
voice translation-rule 24
rule 1 /441612413840/ /3840/
rule 2 /441612413841/ /3841/
rule 3 /441612413842/ /3842/
rule 4 /441612413843/ /3843/
rule 5 /441612413844/ /3844/
rule 6 /441612413845/ /3845/
rule 7 /441612413846/ /3846/
rule 8 /441612413847/ /3847/
rule 9 /441612413848/ /3848/
rule 10 /441612413849/ /3849/
!
voice translation-rule 30
rule 1 // /0/
!
voice translation-rule 410
rule 1 /^9\(.......)\)$/ /441\1/
rule 2 /^9\(.*\)/ /\1/
!
voice translation-rule 1111
rule 1 /\(....\)/ /44161241\1/
rule 2 /\(....\)/ /44161241\1/
rule 3 /\(....\)/ /44161241\1/
rule 4 /3849/ /441612413849/
!
voice translation-rule 1112
rule 1 /^9/ //
!
voice translation-rule 2001
rule 1 /441612413360/ /6000/
!
voice translation-rule 2222
!
!
voice translation-profile CallBlocking
translate called 2222
!
voice translation-profile FAX_Called
translate called 4
!
voice translation-profile MainDDI_Called
translate called 24
!
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate calling 1111
translate called 1112
!
voice translation-profile PSTN_CallForwarding
translate redirect-target 410
translate redirect-called 410
!
voice translation-profile PSTN_Outgoing
translate calling 1111
translate called 1112
translate redirect-target 410
translate redirect-called 410
!
voice translation-profile add0
translate calling 30
dial-peer voice 1000 voip
description ** Incoming call from SIP trunk **
translation-profile incoming add0
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1001 voip
description ** Outgoing call to SIP trunk (Generic SIP Trunk Provider) **
translation-profile outgoing PSTN_Outgoing
destination-pattern 0T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1002 voip
corlist outgoing call-local
description ** star code to SIP trunk **
destination-pattern *..
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
dial-peer voice 1004 voip
description description ** Outgoing call to SIP trunk (Generic SIP Trunk Pro
translation-profile outgoing PSTN_Outgoing
destination-pattern 9T
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
!
thanks
Ian
02-13-2009 07:15 AM
Hi, as Nick suggested, do the debug to check call is hitting the right DP (Dial Peer).
02-16-2009 02:51 AM
Guys
I got this working using:
voice translation-rule 30
rule 1 /^/ /0/
Then adding this to my main incoming DDI dial-peer
Thanks very much for your input
Ian
02-16-2009 02:55 AM
Interesting. For my curiosity only, could you confirm if // /0/ works as well ?
02-16-2009 04:10 AM
yep, that one also works
thanks
Ian
02-16-2009 04:27 AM
Thank you, I've rated your post as you took further time on an already resolved issue.
02-13-2009 05:57 AM
Try this configuration:
voice translation-rule 4
rule 1 /^\(.*\)$/ /0\1/
voice translation-profile add-zero
translate calling 1
dial-peer voice 1 voip
incoming called-number .
translation-profile incoming add-zero
You may want to 'debug voip dialpeer' to see if you're hitting the dial peer you expect to.
-nick
02-13-2009 06:04 AM
Nick, you have rule 4 called as 1 from profile.
The regexp isn't necessary for prepend. // matches the "virtual nothing" in front of number.
02-13-2009 11:27 AM
This is probably correct. I'm kind of a stickler because I too many stupid things happen because of assumptions, so I generally hard code things like this so there isn't any room for error.
-nick
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