[asterisk-users] Asterisk and WebIntegration

[Kurian Thayil]

Hi Geriant,

My apologies for the delay in reply. We won't be using php but Perl and
there is an AGI module for perl Asterisk::AGI. I may be using Manager API
for sending Hangup signal. Im planning to write a bash script which perl
invokes when hangup button is pressed in the web interface. Bash script
telnets and sends Hangup signal to the manager API. I am not yet able to
acheive sending commands via bash script using telnet. But I am trying.

One thing that's confusing me is if in future, incoming facility needs to be
activated and since Auto Answer feature in EyeBeam/Twinkle is ON, don't you
think that would be a problem? I think for that, the possible work around
will be using 2 softphones, say EyeBeam and Xlite together in the same PC.
Configuring one extension in EyeBeam to make outbound calls (with Auto
Answer enabled) and configuring Xlite with an extension which receives
inbound calls. Do you have any suggestion on that?

Regards,

Kurian Mathew Thayil.


On Tue, Mar 10, 2009 at 7:32 PM, Geraint Lee <geraint at gmail.com> wrote:

> 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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