[Asterisk-Users] TDM400P FXO channel hookstate always "Offhook" &outbound digits sent before provider dialtone

Manny A. Wise asterisk-users at calltheus.com
Fri Aug 12 14:16:17 MST 2005


That's a good advice, BUT!!!!
Remember we talking *@home here, it will get overwritten every time you do a
configuration reload...
That is what I did for my cellsocket, but guess what, I had to fix it every
time I mess around with the system even adding an extension, it was soooo
annoying, that now I have two systems running, one with a at h and another
plain *... ;)   Heck at least I learn to deal with both now.. jejejejeje

Manny

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Tom Rymes
Sent: Friday, August 12, 2005 4:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] TDM400P FXO channel hookstate always "Offhook"
&outbound digits sent before provider dialtone

Open the A at H AMP interface, click on trunks, and click on the entry  
for your ZAP trunk. Then, put 'ww' in the "Outbound Dial Prefix"

Tom

On Aug 12, 2005, at 1:29 PM, Stephen Joyce wrote:

> I have an Asterisk at Home 1.3 server (Asterisk 1.0.9) and recently  
> TDM400P with (1) FXO card on port 4. Inbound calls are always  
> but outbound calls fail 75% of the time with intercept messages  
> dial tone provider that include "we're sorry, your call did not go
> through", and "we're sorry, when placing a local call it is now
> necessary to dial an area code and the 7-digit number".
> I have connected a test set in monitor mode to the phone line to  
> to what's being sent out the line by the Zap channel and 10 digits are
> sent but the first digit is usually sent only as I hear the dial tone
> being drawn from the line, so it appears that it's sent before the
> provider is ready to receive it. I can't find any sort of setting that
> would allow me to manually configure a dialing delay on the line,  
> suspect this would provide a band-aid.
> When looking at the Asterisk CLI, I see that the correct number is  
> dialed by my dial plan. I am calling from SIP extension 1100 and  
> 770-555-1234, which is a local 10-digit phone number.
> -- Executing Dial("SIP/1100-9adc", "ZAP/g0/7705551234") in new stack
> -- Called g0/7705551234
> -- Zap/4-1 answered SIP/1100-9adc
> The status of the channel is "Offhook" regardless of whether or not  
> phone line is actually Offhook or completely idle. I'm assuming that
> when the line seems idle, it should show as Onhook.





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