Incoming DID not accepted

FreePBX Tips and Tricks
1

Hi,

Have freepbx installed and running.
Cannot get the internal route to take my DID number.
have one number called 31XX2629301
If I fill that in at DID number it does not accept the call
If I keep the DID number empty then it works, but I want to connect more DID’s to the system and they need to go to their specific extension.

Any ideas ?

the sip trunk says
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: sip:91.223.66.21;lr;ftag=53BA935C-22;did=152.8d4
Date: Sat, 14 Sep 2013 08:09:31 GMT
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: sip:[email protected];tag=53BA935C-22
Allow-Events: telephone-event
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3514141683-474681827-3221028898-2438473394
Timestamp: 1379146171
Content-Length: 383
User-Agent: Cisco-SIPGateway/IOS-12.x
To: sip:[email protected]
Contact: sip:[email protected]:5060
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: [email protected]
Via: SIP/2.0/UDP 91.223.66.21;branch=z9hG4bK421f.831a4eb2.0
Via: SIP/2.0/UDP 91.223.66.32:5060;x-route-tag="cid:[email protected]";branch=z9hG4bKF72C4954D
CSeq: 101 INVITE
Max-Forwards: 10

This should work right ??

Martin

Tips welcome.

2

You have to put an underscore. IE _44455577XX to use expressions in inbound routes.

See tool tip in module for complete details or check the wiki on the module.

3

Oke, I will put the full number. 31702629301 is in the DID Number on inbound routes

The result is still the same, if I call the number from somewhere else, then it doesnt work.

PS. I checked the wiki, and all it says is to put the DID number.
I did as the DID is the same (copy and paste) from the SIP header)

martin

4

Well that’s ccsip debug so doesn’t tell me much.

Need to see the other side.

Basics, is the trunk context from-trunk?

5

Yes,
below the settings of the trunk (under peer).

disallow=all
allow=alaw&ulaw&g729
host=sip.esvoip.nl
fromdomain=sip.esvoip.nl
username=31702629301
type=from-trunk
secret=xxxxxxxxx
fromuser=31702629301
qualify=yes

6

Trunk settings don’t tell me much either. I need Asterisk debug of failed call.

7

I wonder if you are receiving the digits you think you should. If you are not asterisk will speak the actual digits received before dropping the call.

8