[asterisk-users] Asterisk as a Operator Phone

[Kyle Kienapfel]

On Tue, Jul 26, 2011 at 12:37 AM, Nikhil <d.nikhil at cem-solutions.net> wrote:

>
> I am using asterisk as a client not as a server. For client I need features
> like transfer ,call forward ,multiple lines as in normal IP Phones like
> CISOC,polycom.
>
> In asterisk ,we have chan_alsa driver that will communicate to the local
> soundcard. If I installed asterisk in my ubuntu system,and using CLI command
> I can make calls outside and once call connected I can hear and talk from my
> Headphone.
>
> I planing to enhance chan_alsa module to get the features same as in  SIP
> client.
>
> Thanks
> Nikhil
>
>
> On 07/26/2011 12:57 AM, Duncan Turnbull wrote:
>
>> Asterisk can run operator phones with no problem, there are multiple
>> phones out there with addon buttons for automating shared line appearances
>> forwards and other functions
>>
>> For example yealink have the t38 with 6 lines and 16 buttons and the ex 38
>> with 38 additional programmable buttons to add to that if you need
>>
>> Are you talking about a phone that is not sip based?
>>
>> I am not sure why you need to use chan_alsa?
>>
>> Cheers Duncan
>>
>> Sent from my iPhone please excuse the typos
>>
>> On 25/07/2011, at 12:30 AM, Nikhil<d.nikhil at cem-solutions.**net<d.nikhil at cem-solutions.net>>
>>  wrote:
>>
>>  Any reply on this......
>>>
>>> On 07/22/2011 12:56 PM, Nikhil wrote:
>>>
>>>> Hi
>>>>    Does anyone used asterisk as a operator phone,with multiple lines and
>>>> features like transfer forward and etc.I used chan_alsa driver to make
>>>> asterisk as SIP Phone,but it has limitation,we cant make or receive multiple
>>>> calls,and will not able to do any features like transfer forward etc. Is any
>>>> other application available in asterisk to do this .
>>>>
>>>> Thanks
>>>> Nikhil
>>>>
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>>>>
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>>                http://www.asterisk.org/hello
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>>
>
> --
>
> ______________________________**______________________________**_________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>              http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>  http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users>
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Using asterisk as a client sounds interesting. I guess all the existing sip
clients suck?

-Kyle
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