JimNadeau
asked on
FreePBX Asterisk Cisco ATA-187 Unable to make calls
I am trying to get a Cisco ATA-187 to work with FreePBX. The ATA will register and be able to receive calls, but can't make calls. As soon as I dial any number through the ATA I get a busy signal that starts within a second of the first number being dialed.
All of our other phones are Cisco 7940s and 7960s and work just fine.
I am new to FreePBX and any help will be appreciated.
All of our other phones are Cisco 7940s and 7960s and work just fine.
I am new to FreePBX and any help will be appreciated.
ASKER
How do I enter the dialplan?
use a browser and go to the menu item on the ATAs GUI.
ASKER
Below is the only GUI for the ATA I know of and I do not see where to enter a dailplan:
ATAGUI.bmp
ATAGUI.bmp
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ASKER
I understand now...
You are talking about the ATA 186.
There is no /dev for the 187. The 187 gets it's dialplan from an XML on the TFTP.
I enabled SSH on the 187 and captured the log from trying to dial a number, and I should be able to figure it out from there.
I grabbed an 186 from the graveyard and set it up. It works. Putting ATA 187 on back burner for now and deploying ATA 186
You are talking about the ATA 186.
There is no /dev for the 187. The 187 gets it's dialplan from an XML on the TFTP.
I enabled SSH on the 187 and captured the log from trying to dial a number, and I should be able to figure it out from there.
I grabbed an 186 from the graveyard and set it up. It works. Putting ATA 187 on back burner for now and deploying ATA 186
Sorry,
As I said it's been awhile.
In the XML file you should see this line -- <dialTemplate>dialplan.xml </dialTemp late>
What is in your dialplan.xml --
It's the template Match that is the magic -
Look at the DIAL PLAN NOTES notes HERE -
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
As I said it's been awhile.
In the XML file you should see this line -- <dialTemplate>dialplan.xml
What is in your dialplan.xml --
It's the template Match that is the magic -
<?xml version="1.0" encoding="UTF-8"?>
<DIALTEMPLATE>
<TEMPLATE MATCH="..." Timeout="1" User="Phone"></TEMPLATE>
<TEMPLATE MATCH="...." Timeout="2" User="Phone"></TEMPLATE>
<TEMPLATE MATCH="......." Timeout="1" User="Phone"></TEMPLATE>
<TEMPLATE MATCH="1.........." Timeout="1" User="Phone"></TEMPLATE>
<TEMPLATE MATCH="*" Timeout="15"></TEMPLATE>
</DIALTEMPLATE>
Look at the DIAL PLAN NOTES notes HERE -
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+7970+SIP
ASKER
This is the dailplan that I am using for testing and it goes to busy after you dial the first digit.
<?xml version="1.0" encoding="utf-8"?>
<DIALTEMPLATE>
<TEMPLATE MATCH="*" Timeout="10" />
</DIALTEMPLATE>
ASKER
Fixed IT!!!!!
I added a new dial plan just for the ATAs. Called it atadialplan.xml. In the config file for the ATA I pointed them to the new dial plan. In the dialplan I did not have any lines with * or a # and it worked.
I added a new dial plan just for the ATAs. Called it atadialplan.xml. In the config file for the ATA I pointed them to the new dial plan. In the dialplan I did not have any lines with * or a # and it worked.
<?xml version="1.0" encoding="utf-8"?>
<DIALTEMPLATE>
<TEMPLATE MATCH="1.." TIMEOUT="3" />
<TEMPLATE MATCH="2.." TIMEOUT="3" />
<TEMPLATE MATCH="4.." TIMEOUT="3" />
<TEMPLATE MATCH="......." TIMEOUT="5" />
<TEMPLATE MATCH=".........." TIMEOUT="5" />
<TEMPLATE MATCH="1.........." TIMEOUT="5" />
</DIALTEMPLATE>
I will play with it some more and try to see what it is it does not like.
http://www.manualslib.com/manual/27753/Cisco-Ata-186.html?page=108