Incoming call goes to wrong extension

I’m running Asterisk PBX 10.7.0 on centos 5 and I have two phone numbers with draytel.org, both on different usernames and passwords. One number (which I’ve used for years) has the incoming calls going to ext 17, and today I set up a new second phone number and I want it to go to ext 14 but incoming calls on this new number keep going to ext 17.
In the asterisk web GUI the incoming call rule for the new number clearly shows the destination being ext 14. I’ve tried restarting asterisk but it made no difference. And in extensions.conf it also looks ok, this is the entry in it:

[DID_draytel.org] <---- this was my first draytel sip trunk.
include = DID_draytel.org_default
[DID_draytel.org_default]
exten = s,1,Goto(default,17,1)

[DID_]
include = DID_default
[DID
<new-trunk)_default]
exten = s,1,Goto(default,14,1)

I can’t work out why incoming calls on the new trunk keep going to ext 17. I’ve been on this all day and getting nowhere so thanks for any help.

Probably because they are coming from the same IP address and they have no DID number (the extension in the incoming call) to distinguish them.

thankyou for your input david. Is there anything I can do about this? I can’t control the ip address the calls are coming from, but can I do something about the DID number you mentioned? or is that in draytels hands?

You can try specifying a callback number in the register, but if this were true DID rather than VoIP use of the term, they would have provided at least last few digits of the called number in the incoming INVITE.

in order to specify a callback number in the register as you say, which file do I need to edit to achieve this? or can it be done in the web gui? I’m sorry but I’m not familiar with the register in asterisk.

No web GUI is supported on these forums. If you want to ask questions about the capabilities of the web GUI you need to ask the people who support that GUI.

ok so forgetting the web gui… which file do I need to edit to specify a callback number in the register?

In the register line in the SIP configuration, probably sip.conf