[asterisk-users] Asterisk and WebIntegration

[Kurian Thayil]

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)
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090310/e2127ce7/attachment.htm