Follow me => to cell phone not working

Hello,

I would like to do a simple thing but it is not working. I guess the setup is not proper but I don’t know where could be the trick.
When all user phone to the ****20 it is redirected to an user office (user 650) and it’s ringing. This is OK. But I want to make a permanent transfer to his cell phone *****10.

SO, I’m using the follow me option :
Status : Enable
Dialplan : Regular
Destinations by Dial Outside Number : Cell phone number (starting by # or the number itself).

It is not working and I wonder why. If you have a clue or any ideas, you’re welcome.
Thanks,
Yohan

PS : sorry for my poor english.

EDIT 1 : It is working after 30/35seconds.

This is not part of an Asterisk configuration! I guess it may be some GUI, in which case ask on the forum for the GUI, or it might be something on the phone.

Thanks for your answer david55, I’ll do it.

I notice than sometimes the transfert on the cell phone is not working, no reason. Here is the log :

[quote] == Everyone is busy/congested at this time (1:0/0/1)
– Executing [1-dial@macro-trunkdial-failover-0.3:2] GotoIf(“Local/CELL-PHONE@DLPN_SOCIETY-000002c1;2”, “22 > 0 ?1-CHANUNAVAIL,1:1-out,1”) in new stack
– Goto (macro-trunkdial-failover-0.3,1-CHANUNAVAIL,1)
– Executing [1-CHANUNAVAIL@macro-trunkdial-failover-0.3:1] Dial(“Local/CELL-PHONE@DLPN_SOCIETY-000002c1;2”, “SIP/trunk_1/CELL-PHONE”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/trunk_1/CELL-PHONE
– SIP/trunk_1-00002675 is making progress passing it to Local/CELL-PHONE@DLPN_SOCIETY-000002c1;2
– Local/CELL-PHONE@DLPN_SOCIETY-000002c1;1 is making progress
– Got SIP response 603 “Declined” back from @IP_OFTHE_TRUNK_SIP:5060[/quote]

Also, I notice a problem for the “macro” to be link up.
First macro with voice "You have an incoming call"
Second macro with voice : "To accept this call, please tip on STOP "
And the whole macro go on and on and on. I don’t know on what digit I should press…after trying it’s “9” but it’s weird.

PS : A moderator can move my post the Asterisk GUI forum ?

The trunk provider refused the call for reasons other than authentication. You need to ask them.

The other problems should be taken up with the people who wrote the dialplan. That dialplan is not part of Asterisk.