cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
431
Views
0
Helpful
5
Replies

CME Call Forwarding External to External Gets Busy

DarkStormEd
Level 1
Level 1

Hi everyone, trying to resolve a problem. Hoping someone can help. 

We recently started using call forwarding for some of our sales and marketing team when they travel.  I added CFwdAll button and they put in 9 plus cell number.  If I make a test call from an internal extension, it forwards and dials out fine.  If I have a customer calling in from external and then being routed to them through the AA either by prompt of DXD the external caller is getting a busy signal.  Only thing I could find online was possibly our PSTN supplier is blocking the call because the CME is forwarding the CallerID from the external phone and it is obviously not one of our resisted numbers.  Anyone have experience with this?  Solved maybe with a translation on the caller ID somehow?

CME 12.6 on ISR4331 IOS XE 17.02.01r

Thanks

Ed

 

5 Replies 5

All ISP either drop the call or translate the calls to your DID pilot number if you send the ANI other then your DID range. they does this for the Billing purpose.

In your case, for the forwarded calls  the calling information send to ISP will be the mobile number of the person who called. And ISP will be drooping the call.

you can see this on debugs.

You need to apply some translation to convert this to your DID number.



Response Signature


Thank you, that is what I thought it might be.  I will figure out the caller ID translation and hopefully that will fix it.

What is your setup? Are you using ISDN or a SIP trunk?

SIP Trunks.

Then you are probably not allowed to send a number in the FROM header, which is not your assigned access number. Without having a debug output of your call, but I bet you get something like a "403" error code.

You have ordered "CLIP no screening" for that Trunk?
Have you talked to the provider, how such scenarios should work? Normally, you need to send a number of your DID range in the PAI or PPI header. You can do that with SIP profiles. Every SIP trunk I have made so far uses one of those 2 headers for call verification.