First of all i’d like to thank you in advance for the great support in this forum you help until now solves alot of my problems .
Now i make some fresh installs for testing and i find out that all the times when i make a call from an internal number to another it takes 2-3 sec for the destination number to response .
I try also several Grandstream devices GXP200 , GWX4008 , BT100 .
Is there something a can adjust to reduce this dead time ?
IP phones have a dial plan within them. It tells them, based on the first dialed digit that you touch, how many more digits it needs before it should send the dialed digits to the Asterisk system.
If you don’t have a dial plan setup correctly, you usually have to wait for a timeout where the phone will simply send the digits it’s already received.
You can usually get around this by touching a “dial” button on the phone, (on the Grandstream’s I think it’s marked “send”) or sometimes, it’s the “#” key. Grandstream will let you designate the # key as an “end of dialing” indicator.
Check your phone’s documentation for setting up an appropriate dial plan.
Thanks alot my friend
I didnt know that about the dial plan in GXW-4008
I search around that and i set the dial plan as folows
{ 9xxxxx | 9xxxxxxxxxx | 0xxxxxxxxxxxxxx | 3xx | 8xxx }
leading numbers with 9 can be 5 or 10 digits (services & in country calls)
0 for international calls 14 digits
3 for internal calls
8 for the conference rooms
everything works perfect now .
But the GXP2000 dont have any dial plan field in the configuration page .
Yes. The GXP-2000 will require you to use the “send” button located next to the “speaker” button.
You can also choose to make the “#” button indicate end of dialing. (Personally, I would not.)
So, users with a GXP-2000 will need to dial extension 321 as:
3 2 1 SEND
Alternatively, you can go into the “Account” settings page, and set the phone to use “early dial” for the Asterisk account.
This has the effect of touching the “Send” button after each digit is pressed. Each time you dial another number, all previously dialed numbers, plus the new number are sent to the Asterisk system.
So if a user were to dial extension 321 the communications would be like this:
3 :from phone
Code 484 no user “3”:from Asterisk
32 :from phone
Code 484 no user “32”: from Asterisk
321 : from phone
Ok. calling station 321 :from Asterisk
You get a lot of invite messages that are invalid that way, but it will make it so you don’t have to manage the dial plan in the phone as well as in the Asterisk box, and users don’t have to press send or wait for timeouts. As soon as you finish dialing a number that matches something in your Asterisk dial plan, call processing will begin.