UC500 incoming digit prefix

Answered Question
Feb 13th, 2009

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


Correct Answer by Paolo Bevilacqua about 8 years 2 weeks ago

Thank you, I've rated your post as you took further time on an already resolved issue.

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Overall Rating: 5 (4 ratings)
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Paolo Bevilacqua Fri, 02/13/2009 - 04:39

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

ianwarb Fri, 02/13/2009 - 04:49

I will try this, thank you very much for being so prompt

ianwarb Fri, 02/13/2009 - 05:52

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?

Paolo Bevilacqua Fri, 02/13/2009 - 05:57

Possibly it is not matching the DP.

Do you have a single voip DP with incoming called-number ?

ianwarb Fri, 02/13/2009 - 06:12

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


ianwarb Mon, 02/16/2009 - 02:51

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

Correct Answer
Paolo Bevilacqua Mon, 02/16/2009 - 04:27

Thank you, I've rated your post as you took further time on an already resolved issue.

Nicholas Matthews Fri, 02/13/2009 - 05:57

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

Paolo Bevilacqua Fri, 02/13/2009 - 06:04

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.

Nicholas Matthews Fri, 02/13/2009 - 11:27

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

Paolo Bevilacqua Fri, 02/13/2009 - 11:30

Nick, it's not an assumption, it works and it makes sense too. Try it once and you will have less chances of typing wrong with all the funny characters in a fully specified match and set rule.

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