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Calls being forwarded are adding number for accessing outside line

EdLaw
Level 1
Level 1

Hi All,

Trying to understand why this is happening and where to fix it.

 

Example: An external caller calls my desk, its forwarded to my cell phone and the caller ID shows the callers number with an 8 in front of it. This is the number to make an external call if I am sitting at my desk.

 

I appreciate the insight!

 

-Ed

1 Accepted Solution

Accepted Solutions

This is what you have today that would affect called or calling number in or outbound to ATT.

Used in the outbound direction in dial-peer 201-204

voice translation-rule 301
rule 1 /^222/ //

voice translation-profile OUT_TO_ATT
translate called 301

 

Used in the inbound direction in dial-peer 200


voice translation-rule 2
rule 1 /^214.......$/ /8\0/
rule 2 /^469.......$/ /8\0/
rule 3 /^972.......$/ /8\0/
rule 4 /^682.......$/ /8\0/
rule 5 /^..........$/ /81\0/
rule 6 /.*/ /8011\0/

voice translation-profile IN_FROM_ATT
translate calling 2
translate called 200

Voice translation-rule 2 adds an 8 in-front of rule 1 to 4 and 81 in-front of what it match in rule 5 and then 8011 in-front of what it match in rule 6. This is what make the 8 appear to the calling number for calls inbound to you.

What you need to do is remove the 8 for calling number when you send it out to ATT. That can be done by adding below or a variation of it depending on your needs.

voice translation-rule 302
rule 1 /^8\(.*\)/ /\1/

voice translation-profile OUT_TO_ATT
translate calling 302

To be frank this translation is very wide, it will remove an 8 from the start of any calling number that leaves the gateway to ATT. So you might want to make it more specific that what I have in the example.



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View solution in original post

12 Replies 12

Ritesh Desai
Spotlight
Spotlight

Hello @EdLaw 

 

In CUCM Administration / route-patterns, please check if your using predigit "8" in your case to strip 8 and route calls to voice gateway without 8. Second in voice router (VG / CUBE) to strip 8.

 

But one thing is confusing to me, you said call is forwarded to your mobile number which means telecom provider is routing calls. If you think its not correct then you need to check your dial-pattern from CUCM and VOICE ROUTER as well.

 

regards,

Ritesh Desai

*** Please rate helpful post. Please mark as answer if it solves your problem/query.
regards, Ritesh Desai

Sounds like you are rewriting calling number inbound to have an 8 in-front of it to be able to call back to it from your directories list on you phone. Then when you forward the call out to your mobile your keeping that calling number as is when sent out to PSTN. You would need to again rewrite the calling number on the way out to PSTN to remove the 8 that you added in the inbound direction.

 



Response Signature


Thanks! Is this done on the Phone Configuration or somewhere else?

Depending on your country's numbering plan it might be easiest to do it on the voice gateway.  The only issue would be if any of your normal outbound calls present a calling number starting "8".   If they don't then a voice translation rule removing 8 from the beginning should do it.

Thanks for the assistance. I looks like I am too much of a beginner to find this on the gateway. I see on the route pattern configuration under Called Party Transformations that discard digits is set to PreDot. I assumed this would accomplish removing the (8) in front of the number. Am I incorrect?


@EdLaw wrote:

Thanks for the assistance. I looks like I am too much of a beginner to find this on the gateway. I see on the route pattern configuration under Called Party Transformations that discard digits is set to PreDot. I assumed this would accomplish removing the (8) in front of the number. Am I incorrect?


That would affect the called number, not the calling. On the gateway in IOS you would look for voice translation rules. Have alook at this excellent document that explains how this works. https://www.cisco.com/c/en/us/support/docs/voice/call-routing-dial-plans/61083-voice-transla-rules.html

 



Response Signature



@EdLaw wrote:

Thanks for the assistance. I looks like I am too much of a beginner to find this on the gateway. 


The translation rule that I suggested would be configured on the gateway itself, not in the gateway settings in Callmanager.

Hi,

Here is some of the config from the VG. Does this show where are we adding the (8) to calls sent to the phones and is there a rule I can add to strip the (8) on calls being forwarded to their external phones? Do you need more of the config? Thanks again!

 

multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
trunk group PSTN
!
!
!
voice-card 0
dsp services dspfarm

!
!

allow-connections sip to sip
fax protocol t38 nse version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw
h323
call preserve limit-media-detection

!
!

voice translation-rule 1
rule 1 /^8\(011.*\)/ /\1/ type any unknown plan any unknown
!
voice translation-rule 2
rule 1 /^214.......$/ /8\0/
rule 2 /^469.......$/ /8\0/
rule 3 /^972.......$/ /8\0/
rule 4 /^682.......$/ /8\0/
rule 5 /^..........$/ /81\0/
rule 6 /.*/ /8011\0/
!
voice translation-rule 3
rule 1 /^4700$/ /214373\0/
rule 2 /^[467]...$/ /214706\0/
!
voice translation-rule 200
rule 1 // /111/
!
voice translation-rule 201
rule 1 /^111/ //
!
voice translation-rule 300
rule 1 // /222/
!
voice translation-rule 301
rule 1 /^222/ //
!
!
voice translation-profile IN_FROM_ATT
translate calling 2
translate called 200
!
voice translation-profile IN_FROM_UCM
translate called 300
!
voice translation-profile OUTBOUND-INTERNATIONAL
translate called 1
!
voice translation-profile OUT_TO_ATT
translate called 301
!
voice translation-profile OUT_TO_CM
translate called 201
!
voice translation-profile PSTN-INBOUND
translate calling 2
!
voice translation-profile SRST-ANI
translate calling 3
!
!

dial-peer voice 1 pots
translation-profile incoming PSTN-INBOUND
shutdown
incoming called-number [467]...$
direct-inward-dial
!
dial-peer voice 2 pots
trunkgroup PSTN
description Ten Digit Local Dialing
translation-profile outgoing SRST-ANI
shutdown
destination-pattern 8[2-9].........
forward-digits 10
!
dial-peer voice 3 pots
trunkgroup PSTN
description National Long Distance Dialing
translation-profile outgoing SRST-ANI
shutdown
destination-pattern 81[2-9].........
forward-digits 11
!
dial-peer voice 4 pots
trunkgroup PSTN
description International Long Distance Dialing
translation-profile outgoing OUTBOUND-INTERNATIONAL
shutdown
destination-pattern 8011T
forward-digits all
!
dial-peer voice 100 voip
tone ringback alert-no-PI
shutdown
destination-pattern [467]...
progress_ind setup enable 3
session target ipv4:x.x.x.x
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 5 pots
trunkgroup PSTN
description Emergency Services
translation-profile outgoing SRST-ANI
shutdown
destination-pattern 8?911
forward-digits 3
!
dial-peer voice 200 voip
description Incoming From ATT IPFlex
translation-profile incoming IN_FROM_ATT
session protocol sipv2
incoming uri via ipflex
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 201 voip
description Outgoing From ATT IPFlex
translation-profile outgoing OUT_TO_ATT
destination-pattern 222.+
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 202 voip
description Outgoing From ATT IPFlex
translation-profile outgoing OUT_TO_ATT
preference 2
destination-pattern 222.+
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 203 voip
description Outgoing From ATT IPFlex
translation-profile outgoing OUT_TO_ATT
preference 3
destination-pattern 222.+
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 204 voip
description Outgoing From ATT IPFlex
translation-profile outgoing OUT_TO_ATT
preference 4
destination-pattern 222.+
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/1
voice-class sip bind media source-interface GigabitEthernet0/1
dtmf-relay rtp-nte
no vad
!
dial-peer voice 301 voip
translation-profile outgoing OUT_TO_CM
preference 1
destination-pattern 111.+
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
!
dial-peer voice 300 voip
description Incoming from UCM
translation-profile incoming IN_FROM_UCM
session protocol sipv2
incoming uri via ucm
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
no vad
!
dial-peer voice 302 voip
translation-profile outgoing OUT_TO_CM
preference 2
destination-pattern 111.+
session protocol sipv2
session target ipv4:x.x.x.x
voice-class codec 1
voice-class sip bind control source-interface GigabitEthernet0/0
voice-class sip bind media source-interface GigabitEthernet0/0
dtmf-relay rtp-nte
!
!
!
!
gatekeeper
shutdown
!

 

This is what you have today that would affect called or calling number in or outbound to ATT.

Used in the outbound direction in dial-peer 201-204

voice translation-rule 301
rule 1 /^222/ //

voice translation-profile OUT_TO_ATT
translate called 301

 

Used in the inbound direction in dial-peer 200


voice translation-rule 2
rule 1 /^214.......$/ /8\0/
rule 2 /^469.......$/ /8\0/
rule 3 /^972.......$/ /8\0/
rule 4 /^682.......$/ /8\0/
rule 5 /^..........$/ /81\0/
rule 6 /.*/ /8011\0/

voice translation-profile IN_FROM_ATT
translate calling 2
translate called 200

Voice translation-rule 2 adds an 8 in-front of rule 1 to 4 and 81 in-front of what it match in rule 5 and then 8011 in-front of what it match in rule 6. This is what make the 8 appear to the calling number for calls inbound to you.

What you need to do is remove the 8 for calling number when you send it out to ATT. That can be done by adding below or a variation of it depending on your needs.

voice translation-rule 302
rule 1 /^8\(.*\)/ /\1/

voice translation-profile OUT_TO_ATT
translate calling 302

To be frank this translation is very wide, it will remove an 8 from the start of any calling number that leaves the gateway to ATT. So you might want to make it more specific that what I have in the example.



Response Signature


Thanks @Roger Kallberg ! This is a huge help. If I added translate calling 302 in your example, how would that affect translate calling 2?


@EdLaw wrote:

Thanks @Roger Kallberg ! This is a huge help. If I added translate calling 302 in your example, how would that affect translate calling 2?


It would not affect that rule set at all as it’s used on another dial-peer in the inbound direction.

 



Response Signature


There are many places where this could be set. See below for a few, but for sure not a complete list.

  • Gateway in CUCM
  • Trunk, if you use SIP for communication to your voice gateways
  • Calling party transformation, can be set on various places
  • In IOS on voice gateways with voice translation rules
  • Translation pattern
  • Route Pattern
  • Route List details for Route Group

Personally I tend to prefer doing it in IOS as that's the most flexible way of doing it and it's on the ingress/egress border.



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