03-23-2023
01:11 PM
- last edited on
03-24-2023
05:21 PM
by
Jimena Saez
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.
03-25-2023 05:28 AM
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.
03-28-2023 12:42 PM
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.
I also changed the Calling Party Selection to Last Redirected Number (External)
Do you have anything else in mind that I could try?
Thanks again
04-03-2023 07:45 AM
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
05-15-2023 12:38 PM
in this case user needs to route the call to internal number not the external number
05-15-2023 12:44 PM
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
04-06-2023 08:48 AM
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
06-12-2023 10:59 PM
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
09-10-2023 05:02 AM
please check prefix you use it and CSS is true .
12-28-2023 11:26 PM - edited 12-28-2023 11:27 PM
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).
01-18-2024 08:41 AM
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.
01-17-2024 07:24 AM
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
01-17-2024 03:12 PM
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.
01-18-2024 08:40 AM
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.
08-21-2024 12:23 AM
Please use the forward option in the line configured in phone configuration. there multiple call forward option available for external calls like busy external, answer external, coverage external, unregistered external
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