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External calls are not redirected correctly

Translator
Community Manager
Community Manager

Hello,

We are currently using a Be6000 system and when we schedule a call forwarding on our phones (model 7945) to external numbers, it works perfectly when the original call comes from an internal position, but when it is an external call then this call is sent to the voice box of the telephone station on which the reference is scheduled. I would like to see outsiders also transferred and not put in the voice box.

I know how to find myself in the configuration, but I'm not a specialist enough to know where exactly to look. 

Do you have any suggestions? 

Thank you very much for your support. 

call forwarding.png

13 Replies 13

I can see call forward set external to Voice mail. Uncheck the voice mail box and mentioned the number to forward.

If calls from external which forward to an external number  are failing, probably this is due to calling number which is send to ISP. Changing  Last redirecting on gateway could solve it.

NithinEluvathingal_0-1679747265810.png

 



Response Signature


Thanks for your answer Nithin. 

I unchecked every voicemail check box and pasted the external phone number in every fields just to make sure but its still forward me to the voice mail. 

 

Zone3TI_2-1680032374026.png

I also changed the Calling Party Selection to Last Redirected Number (External) 

Zone3TI_0-1680032264017.png

Do you have anything else in mind that I could try? 

Thanks again  

Hi,

No need to change the call forward external as you have already set the Call Forward All to external number. This configure will override the other call forward settings. 

Can you tell us how you are routing the external calls, whether its via PRI or SIP trunk and if possible share the debugs from the gateway for one working (Call from internal number forwarded to external number) and one non working call (Call from external number) to check what is the Calling number being sent to provider.

If you are using PRI, get "debug isdn q931" and if you have SIP trunk, get "debug ccsip messages"

HTH
Rajan
Please mark all useful posts as helpful and solutions as accepted wherever applicable

 

in this case user needs to route the call to internal number not the external number 

this configuration need to check the check following perameters

1.check if any application dial rule applied to discard any dight

2.check for forced authorisation code is applied for the routing

3.check if right css is applied for call forward option is applied

 

Sadav Ansari
VIP Alumni
VIP Alumni

Run the DNA and share the call flow for external number forwarding.

 

Also put the number on call forward all section and provide the appropriate css.

Then check the Route Pattern for that external number if it’s on cucm or not.

If RP is there and pointing toward the correct voice Gateway then collect the debug from voice gateways and share here.

 

debug isdn q931

debug ccsip message

debug ccapi input 

 

++show run

++show dial-peer voice summary

Pls rate if its “Helpful”. If this answered your question then pls click “Accept as Solution”.

 

BR,

Sadav Ansari

Hi

On the line configuration Page in the call forwarding section, change the Calling Search space Activation policy to "With Configured CSS" value and let me know if it makes any difference.

 

HTH

 

Regards

 

Carlo

Please rate all helpful posts "The more you help the more you learn"

ahmed.zakaria
Beginner
Beginner

please check prefix you use it and CSS is true .

Kaloyan Botev
Beginner
Beginner

Check if the outbound diversion header is enabled on the SIP trunk. The most common issue when forwarding external calls back to the PSTN is that the service provider sees a number that doesn't belong to your system originating from it and they reject it. The diversion header authenticates the calls since it contains the number of the phone that forwarded the call (which should be a number from your range/block of numbers).

Either the calling party should show as your main billing number or any number that you have purchased from provider so they can accept it. Its a mechanism to prevent toll fraud.

You should change the numbers below to your main number or any purchased DID so it should be recognized by your provider.

In case you see diversion header being sent, need to modify as below. The only way to change a caller ID on the CUBE is by using SIP PROFILES, which are used to modify SIP headers. In this specific case, to modify the received Caller ID.
voice class sip-profiles <profile number>

request ANY sip-header P-Asserted-Identity modify "sip:(.*)@" "sip:+1757648XXXX@"
request ANY sip-header Diversion modify "sip:(.*)@" "sip:+1757278XXXX@"

Sometimes, we don't see diversion header as call manager only sends it for forwarded calls, we can instead of modifying header, can add the diversion header itself or change the P-asserted identity of the SIP invite.


voice class sip-profiles <profile number>
request ANY sip-header Diversion add "Diversion: <sip:+1757278XXXX@Gateway-IPaddress>"
request ANY sip-header Diversion add "Diversion: <sip:+1757648XXXX@Gateway-IPaddress>"

For exact SIP profile to be created, need gateway debugs.

  • Debug voip ccapi inout
  • Debug ccsip messages
  • Show run
  • Show version


    Regards,
  • Mohammed A.R.A

Translator
Community Manager
Community Manager

I also be interested in seeing the output of debugs on the gateway showing the call coming in from the PSTN and CUCM returning it to the router. It is possible your original config was perfect, but the gateway (or your provider) was rejecting the hairpinned call.

Maren

I would also be interested to see the debug output on the gateway showing the incoming call of the PSTN and CUCM returning it to the router. Your original configuration may be perfect, but the gateway (or your provider) may reject the pinned call.

Maren

Mohammed-A.R.A
Beginner
Beginner

This can be achieved by either modifying or adding headers.
Sometimes we need to modify headers P-Asserted, Remote-Party and Diversion based on received messages. In cases, where we are not seeing diversion header, need to add it.

 

 

Mohammed-A.R.A
Beginner
Beginner


Either the calling party should show as your main billing number or any number that you have purchased from provider so they can accept it. Its a mechanism to prevent toll fraud.

You should change the numbers below to your main number or any purchased DID so it should be recognized by your provider.

In case you see diversion header being sent, need to modify as below. The only way to change a caller ID on the CUBE is by using SIP PROFILES, which are used to modify SIP headers. In this specific case, to modify the received Caller ID.
voice class sip-profiles <profile number>

request ANY sip-header P-Asserted-Identity modify "sip:(.*)@" "sip:+1757648XXXX@"
request ANY sip-header Diversion modify "sip:(.*)@" "sip:+1757278XXXX@"

Sometimes, we don't see diversion header as call manager only sends it for forwarded calls, we can instead of modifying header, can add the diversion header itself or change the P-asserted identity of the SIP invite.


voice class sip-profiles <profile number>
request ANY sip-header Diversion add "Diversion: <sip:+1757278XXXX@Gateway-IPaddress>"
request ANY sip-header Diversion add "Diversion: <sip:+1757648XXXX@Gateway-IPaddress>"

For exact SIP profile to be created, need gateway debugs.

  • Debug voip ccapi inout
  • Debug ccsip messages
  • Show run
  • Show version


    Regards,
  • Mohammed A.R.A
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