[Asterisk-Users] A unique 'click to call' project - Could usesome advice

Fri Feb 17 11:16:29 MST 2006

Thanks Colin!

Makes sense; I will work on this later today.

If you can, sending the example would be great.

Thanks,

-- -- -- 
 Christopher T. Aloi 
 USA Datanet - Technical Support Engineer 
 318 South Clinton Street 
 Syracuse, NY 13202 
 C: (315) 569 4033 
 O: (315) 579 7074 
 E: caloi at usadatanet.com <mailto:caloi at usadatanet.com>  
-- -- -- 

  _____  

From: Colin Anderson [mailto:ColinA at landmarkmasterbuilder.com] 
Sent: Friday, February 17, 2006 12:36 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could
usesome advice

Same as before but instead of SIP as the origination channel you pass
ZAP/g0/XXXXXXXXXXX (the DID of the agent) to your .call file. In fact,
this is exactly how the www.landmarkhomes.ca script works (it calls the
guy who entered his phone number in the website, when he picks up, it
calls the salesperson's cell number and the two are bridged together)

The drawback is, of course, that it uses 2 ZAP channels to bridge the
call together, but this isn't a problem I guess for you since you seem
to have ZAP channels coming out of your yinyang. 

I have an implementation in Active Server Pages (we are a MS shop) that
I can send you - it's suprisingly simple - but it could be easily
modified for PHP or what have you. 

	-----Original Message-----
	From: Aloi, Christopher [mailto:caloi at usadatanet.com]
	Sent: Friday, February 17, 2006 9:56 AM
	To: Asterisk Users Mailing List - Non-Commercial Discussion
	Subject: RE: [Asterisk-Users] A unique 'click to call' project -
Could usesome advice

	Colin,

	Thanks for your assistance.

	Reading over your advice I seem to still be a bit confused.

	My agents are not on the Asterisk server; it appears in your
advice that my the call will travel this path:

	WWW interface --> agent enters their DID, platform to use, and
termination DID --> AST calls agent --> Agent calls termination DID

	If my agents are not on the Asterisk server (believe me, I wish
there were) :) how will this work?

	I need a way to pass both the desired termination DID and the
origination DID.

	Maybe I missed something....

	Thanks,

	-- -- -- 
	 Christopher T. Aloi 
	 USA Datanet - Technical Support Engineer 
	 318 South Clinton Street 
	 Syracuse, NY 13202 
	 C: (315) 569 4033 
	 O: (315) 579 7074 
	 E: caloi at usadatanet.com <mailto:caloi at usadatanet.com>  
	-- -- -- 

  _____  

	From: Colin Anderson [mailto:ColinA at landmarkmasterbuilder.com] 
	Sent: Friday, February 17, 2006 10:42 AM
	To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
	Subject: RE: [Asterisk-Users] A unique 'click to call' project -
Could usesome advice

	You create a context in your dialplan that accepts the DID to
call as a variable using the SetVar: syntax in your .call file. You then
set up the context to call your agent, and when they pick up, the
context takes the variable you set in your .call file as the dialstring
argument for a subsequent Dial(). Once the DID picks up, the calls are
bridged together. Whatever web scripting language you use writes the
.call file, and you use POSTed arguments or querystrings:

http://foo.com/call?context=MyContext&Agent=SIP/5555&DID=15555551212

	You can see this in action at www.landmarkhomes.ca - click on
any of the pretty buttons that say "Call us now" 

	However, I have noticed that * 1.2.x will not wait for the
caller to pick up before executing the rest of the directives in the
context - it keeps executing regardless of the calling party's pickup
status. Using * 1.0.x the context will wait for the caller to pick up
before placing the call to the callee (i.e. executing the rest of the
directives in the context) 

	.call file (shortened to relevant)

	Channel:     SIP/XXXX (if you are using SIP phones)
	SetVar:    DID=XXXXXXXXXXX 
	Context: MyContext

	[MyContext]
	exten => s,1,Dial(ZAP/g0/${DID})

	hth

		-----Original Message-----
		From: Aloi, Christopher [mailto:caloi at usadatanet.com]
		Sent: Friday, February 17, 2006 8:07 AM
		To: Asterisk Users Mailing List - Non-Commercial
Discussion
		Subject: [Asterisk-Users] A unique 'click to call'
project - Could use some advice

		Hello List,

		I work for an IP communication provider in upstate NY as
the engineer assisting our technical support team.
		We provide a number of different Telco systems to
residential subscribers; and in an effort to more effectively trouble
shoot termination problems I came up with the idea of creating a click
to call system that will allow our agents to effortlessly place test
calls.

		On a daily basis we place numerous (50-100) 'test' calls
to various locations in the US; these 'test' calls are routed using one
of three different phone systems:

		1) The PSTN
		2) Broadband phone platform one
		3) Broadband phone platform two

		I have an Asterisk server configured that can terminate
out three platforms listed above.

		Our support agents are behind a Televantage ACD using
D-TermSeries E NEC phones.  
		Each agent has a DID and are permitted to receive
inbound calls on that DID.

		Here is my goal:

		Create a web application that will allow the agent to
enter the following information into a form:

		1) The agents DID
		2) The platform the agent wishes to terminate a test
call through (either 1,2,3 above)
		3) The number the agent wishes to terminate to 

		My thought is this form will generate a .call file in
/var/spool/asterisk/outgoing that will then ring the agents station,
pause, and terminate to the selected DID using the selected platform.  I
also thought about interacting directly with the AGI.

		I can successfully generate the .call files, and ring a
station on the Asterisk server - the problem is the agents are not on
the Asterisk server.

		Is there a way to use Asterisk to initiate these test
calls?

		Is it possible to create a forwarding context to handle
this?

		Any thoughts?

		Thanks for the help!

		Cheers,

		-- -- -- 
		 Christopher T. Aloi 
		 USA Datanet - Technical Support Engineer 
		 318 South Clinton Street 
		 Syracuse, NY 13202 
		 C: (315) 569 4033 
		 O: (315) 579 7074 
		 E: caloi at usadatanet.com <mailto:caloi at usadatanet.com>  
		-- -- -- 

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