Re: [Freeswitch-users] Newbie question: Why can't I dial?

2009-03-17 Thread Michael Collins
On Tue, Mar 17, 2009 at 3:35 PM, Mark Thomas
 wrote:
> Hello, everyone.
>
> I am new to Freeswitch, and telephony in general.  I am trying to set up a 
> Freeswitch system at work for a project, and I have hit a wall.
>
> I have a dedicated LD T1 from Qwest and a Sangoma A104 card.  I believe I 
> have openzap correctly installed in wanpipe mode. I am trying to bridge an 
> incoming SIP call from an IP phone to an openzap channel without success.  
> The Freeswitch log shows that dialing takes place, but the call never 
> completes.
>
> The call log is here: http://pastebin.freeswitch.org/7805
> The dialplan xml, openzap.conf, and openzap.conf.xml are here: 
> http://pastebin.freeswitch.org/7806
>
> Any help greatly appreciated.
>

Actually I found two things you need to change in the dialplan. What's
happening is that you are telling openzap to dial out span 1, lowest
channel number, but you don't actually give it a phone number to dial.
Here's the current dialplan:

 


  

first, your expression is a bit dangerous. second, it doesn't actually
"capture" the dialed number. I recommend that you do something like
this:



Note the leading nine, the \d+ and the parentheses. Essentially this regex says:
Match any string of digits that begins with a 9 and has at least one
additional digit.
The parens will put the value of (\d+) into the variable $1.

Your bridge then would be this:



Now, reload your dialplan (press F6 or type "reloadxml" at the CLI)
and dial out with a leading 9:
95551212 will send 5551212 to the telco.

Try it and report back!
-MC (IRC: mercutioviz)


> --Mark
>
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Re: [Freeswitch-users] Newbie question: Why can't I dial?

2009-03-17 Thread Brian West

Mark,
	You should join both #openzap and #freeswitch on irc.freenode.net  
there are way too many things to go over and the list would just be  
too slow.


/b

On Mar 17, 2009, at 5:35 PM, Mark Thomas wrote:


Hello, everyone.

I am new to Freeswitch, and telephony in general.  I am trying to  
set up a Freeswitch system at work for a project, and I have hit a  
wall.


I have a dedicated LD T1 from Qwest and a Sangoma A104 card.  I  
believe I have openzap correctly installed in wanpipe mode. I am  
trying to bridge an incoming SIP call from an IP phone to an openzap  
channel without success.  The Freeswitch log shows that dialing  
takes place, but the call never completes.


The call log is here: http://pastebin.freeswitch.org/7805
The dialplan xml, openzap.conf, and openzap.conf.xml are here: 
http://pastebin.freeswitch.org/7806

Any help greatly appreciated.

--Mark

__


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[Freeswitch-users] Newbie question: Why can't I dial?

2009-03-17 Thread Mark Thomas
Hello, everyone.

I am new to Freeswitch, and telephony in general.  I am trying to set up a 
Freeswitch system at work for a project, and I have hit a wall.

I have a dedicated LD T1 from Qwest and a Sangoma A104 card.  I believe I have 
openzap correctly installed in wanpipe mode. I am trying to bridge an incoming 
SIP call from an IP phone to an openzap channel without success.  The 
Freeswitch log shows that dialing takes place, but the call never completes.

The call log is here: http://pastebin.freeswitch.org/7805
The dialplan xml, openzap.conf, and openzap.conf.xml are here: 
http://pastebin.freeswitch.org/7806

Any help greatly appreciated.

--Mark

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