Lord this has been fun to set up.
Had it working before with Asterisk/SARK/SAIL but trying it with FreePBX on my trusty SME Server
First remember that calls go like this :
PSTN > Trunk > Inbound Route > Extension > Outbound Route > Trunk > PSTN
1. Devices
Secret : a password
Device ID : a number e.g start from 9001
Device type : adhoc
This is better as users have to log in and out using *11 and *12 giving you a 2 layer security
2. Users
User Extension : a number e.g. start from 5001
User password : a multi digit PIN
Display Name : username
3. Ring Groups (if required)
Ring-Group Number : pick a number e.g. 600
Description : Draytel Ring all
Extension list : add your extensions
4. Trunk
I have had various settings. This works but I get 'Unmonitored' in Asterisk info and I'm sure it wasn't when I first did it - I thought it said 'OK'. Such is life.
Name : Draytel
Outbound CID : <Your PSTN number>
CID Options : Allow Any CID
Max Channels - Set currently at 2 but I think it can be 5
Outgoing Settings :
Name : Draytel
Peer Details :
type=friend
username=your customer/account number
fromuser=your customer/account number
secret=your customer/account password
host=draytel.org
fromdomain=draytel.org
qualify=yes
context=from-pstn
dtmfmode=rfc2833
insecure=very
disallow=all
allow=ulaw
Incoming Settings : are empty
Registration string :
In the following form
accountnumber:password@draytel.org/your incoming number
e.g. :
6548975:pass@draytel.org/02091234567
5. Outbound Route
Name : Draytel Outbound
Some Dial patterns
Trunk Sequence for Matched : Draytel
6. Inbound Route
Desctiption : Draytel Inbound
DID number : your PSTN Drayel Number
Destination : Pick an extension or a Ring Group
That should be it.
For issues with no sound look carefully at your NAT settings (after you have checked your Mic is working !) I think that 99% of all sound problems are due to incorrect NAT settings, so check, check, and check again.
SIP Phones settings :
Qutecom :
Username : The device name from above e.g. 9001
Password : from the Device password
SIP Domain/Realm : IP or Domain name of server
Displayname : Your name :-)
If necessary
Server : IP or Domain name of server
Proxy: IP or Domain name of server
Enjoy :-)
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