asterisk users - Feb 2006 - 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@usadatanet.com <mailto:caloi@usadatanet.com>  
-- -- -- 
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On 2/17/06, Aloi, Christopher <caloi@usadatanet.com>
wrote:
> 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?
>
 There's a couple ways to go about getting the calls to your agents.
The first, and probably easiest would just be to dial out through the
PSTN on Asterisk to the DIDs on the other system, but that obviously
is a waste cost and resources wise.

 The second, would probably be to do a PRI DS1 tie line between the
NEC and Asterisk and then just dial the DID on a Zap channel that
belongs to that span to get to your agents.

 But your objectives sound like they are definitely achievable with some work.

 BJ

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

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
<http://foo.com/call?context=MyContext&Agent=SIP/5555&DID=15555551212>
&Agent=SIP/5555&DID=15555551212
 
You can see this in action at www.landmarkhomes.ca
<http://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@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:  <mailto:caloi@usadatanet.com> caloi@usadatanet.com 
-- -- -- 

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I'm using the telnet manager interface with the 'originate' command,
just a little perl script that connects and has asterisk dial the
selected number.
It rings the extension first, if they pick up, it'll dial the remote
number.
It's one of the showcase features of the new phonesystem for us :) and
it was surprisingly easy to implement.

-- 
Regards,
Gerard Saraber
Network Admin, Rarcoa, Inc.
(630) 654-2580 x11
gsaraber@rarcoa.com

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 <http://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@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:  <mailto:caloi@usadatanet.com> caloi@usadatanet.com 
-- -- -- 
 

  _____  

From: Colin Anderson [mailto:ColinA@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
<http://foo.com/call?context=MyContext&Agent=SIP/5555&DID=15555551212>
&Agent=SIP/5555&DID=15555551212
 
You can see this in action at www.landmarkhomes.ca
<http://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@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:  <mailto:caloi@usadatanet.com> caloi@usadatanet.com 
-- -- -- 

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<%
language=vbscript 'ASP page
PhoneNumber=Request.Querystring("PhoneNumber") 'Get the number to
call on
the first leg of the call
PhoneNumber2=Request.Querystring("PhoneNumber2") 'Get the number
to
terminate the call to
Context=request.querystring("Context") 'Get the context to execute
on the
Asterisk box
Extension=request.querystring("Extension") 'Optional, start from a
particular extension, otherwise pass the 's' extension
PhonePrefix=Left(PhoneNumber,3) 'Grab out the first 3 digits of the NAMA
number to determine if it's local or LD
'from the perspective of the Asterisk box
If PhonePrefix="780" then 'If it's a local call, then
NumberToCall=Right(PhoneNumber,7) 'Return the rightmost 7 digits
Else
NumberToCall= "1" & PhoneNumber 'It's a LD call, so append
1 to the NAMA
number
End If
Randomize 'We create a pseudo-random number for the .call file filename
set fs = CreateObject("Scripting.FileSystemObject")
set file = fs.CreateTextFile("  <file://\\192.168.1.46\outgoing\>
\\192.168.1.46\outgoing\" & int(Rnd * 1000) & ".call",
true, false) 'Dump
the call file to a SAMBA share on the Asterisk box.
'A looping script running on the Asterisk box looks for any .call files, and
if it finds one, moves the .call file to /var/spool/asterisk/outgoing
'You can do this in PHP with an SCP command instead
file.WriteLine("Channel: ZAP/g0/" & NumberToCall)
file.WriteLine("SetVar: CALLEDPARTY=" & PhoneNumber2)
file.WriteLine("CallerID: 18888988884") 'Set your caller ID where
appropriate
file.WriteLine("MaxRetries: 2")
file.WriteLine("RetryTime: 60")
file.WriteLine("WaitTime: 60")
file.WriteLine("Context: " & Context)
file.WriteLine("Extension: " & Extension)
file.WriteLine("Priority: 1")
file.Close 'Finish up and unlock the file for the looping script to pick up
response.write "&OK=1" 'Response.write a variable back to
Flash indicating
the call was placed successfully. Otherwise you could
'write back some HTML saying "Your call was executed correctly" or
some
such.
%>
 
URL to dial:
 
http://www.foo.com/call.asp?PhoneNumber=7805551212
<http://www.foo.com/call.asp?PhoneNumber=7805551212&PhoneNumber2=7805551213&
Context=MyContext&extension=s>
&PhoneNumber2=7805551213&Context=MyContext&extension=s
 
[MyContext] 'In Asterisk 1.0.X, this context is executed AFTER the calling
party picks up the phone, which is what we want. 
 
exten => s,1,Dial(ZAP/g0/${CALLEDPARTY},40,T)
 
hth

-----Original Message-----
From: Aloi, Christopher [mailto:caloi@usadatanet.com]
Sent: Friday, February 17, 2006 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] A unique 'click to call' project - Could
usesome advice


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:  <mailto:caloi@usadatanet.com> caloi@usadatanet.com 
-- -- -- 

 

  _____  

From: Colin Anderson [mailto:ColinA@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 <http://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@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:  <mailto:caloi@usadatanet.com> caloi@usadatanet.com 
-- -- -- 
 

  _____  

From: Colin Anderson [mailto:ColinA@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
<http://foo.com/call?context=MyContext&Agent=SIP/5555&DID=15555551212>
&Agent=SIP/5555&DID=15555551212
 
You can see this in action at www.landmarkhomes.ca
<http://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@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:  <mailto:caloi@usadatanet.com> caloi@usadatanet.com 
-- -- -- 

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