[asterisk-users] Asterisk and WebIntegration

[Geraint Lee]

If you're using a php i'd take a look at phpagi - there are others around
for various different languages too. our agents use twinkle with
auto-answer, the only reason they need to look at twinkle is if they need to
perform a transfer (that too will soon be done from the web browser), you
can do pretty much anything with the asterisk manager (originate the call
and hangup the call and a load of other useful stuff)

Cheers

2009/3/10 Kurian Thayil <kurianmthayil at gmail.com>

> Hi Steve,
>
> That worked beautifully. Thank you so much. But one question though.
> Imagine if I keep a Hangup Button in the interface and it should terminate
> the call. Will that be possible? This scenario happens when the user gets
> connected to an invalid phone number where the user have to manually
> disconnect. I don't plan to confuse the user by asking them to use eyebeam
> to disconnect the call. If it could be integrated to the web interface they
> just have to stick on to that alone. Is there any way?
>
> Regards,
>
> Kurian Mathew Thayil.
>
> On Tue, Mar 10, 2009 at 4:51 PM, Steve Totaro <
> stotaro at first-notification.com> wrote:
>
>>
>>
>>   On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil <kurianmthayil at gmail.com
>> > wrote:
>>
>>> Hi All,
>>>
>>> Is there a way that I can include call dialing functionality in a
>>> webinterface. I have EyeBeam configured with a SIP user say
>>> 8440. Will I be able to design an inteface which agent can choose a
>>> number and the Dial without punching in the number in
>>> Eyebeam.
>>> I tried using the .call file. ie The agent can choose which number to
>>> dial from a web interface. Then, a .call file is
>>> created with the following informations.
>>>
>>> Channel: Zap/g2/9444204943
>>> Context: inbound_support
>>> Extension: 8440
>>> Priority: 0
>>>
>>> Now, in the extensions.conf file, I mentioned the following under
>>> inbound_support context.
>>>
>>> [inbound_support]
>>> exten =>8440,1,Dial(SIP/8440,55,tTo)
>>> exten =>8440,2,Answer
>>> exten =>8440,3,Hangup
>>>
>>> But, here the call gets connected only when the receiver end receives the
>>> call. When the receiver end picks up the phone,
>>> SIP/8440 rings.
>>>
>>> Is there any other way to implement this. I am not ready to use Vicidial
>>> (AstGUIClient) because the interface to be designed
>>> is too custom and the agent should have the list of numbers in front of
>>> them while they dial which cannot be done using
>>> Vicicial.
>>>
>>> Regards,
>>>
>>> Kurian Mathew Thayil.
>>>
>>
>> The following will ring the internal support personnel (8440) first, after
>> answered, it will then dial the customer (14109850123) (Are you in
>> Maryland?)
>>
>> Turn on auto-answer and it should be seamless.
>>
>>
>> Stolen from Wiki:
>>
>> To create a call to 14109850123 on a SIP phones called bt101, here's the
>> file you'd create in /var/spool/asterisk/outgoing (whatever name is good, of
>> course must be accessible and deletable by asterisk GNU/Linux user):
>>
>> Channel: SIP/8440
>> MaxRetries: 1
>> RetryTime: 60
>> WaitTime: 30
>>
>> #
>> # Assuming that your outgoing call logic is kept in the #  context called [outgoing]
>>
>> # Context: outgoing
>> # Extension: 14109850123
>> # Priority: 1
>>
>>
>> --
>> Thanks,
>> Steve Totaro
>> +18887771888 (Toll Free)
>> +12409381212 (Cell)
>>
>>
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>
>
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