[asterisk-dev] Seeking VOIP / Asterisk Guru for Small Project
Shane R. Spencer
shane at bogomip.com
Thu Jul 28 16:23:31 CDT 2011
/me finds his broadsword.
On 07/28/2011 12:54 PM, Andre Courchesne wrote:
> Sorry that was intended for Brian only, you can go head and cut my head off now...
>
>
> Quoting "Andre Courchesne" <courchea at net-forces.com>:
>
>> Hi Brian,
>>
>> I am outside the office right now but would be happy to help you. I have many Asterisk
>> installation under my belt and can direct you on hardware, carrier and asterisk
>> configuration.
>>
>> I will study your requirements further tonight, but I believe I will need to educate
>> you a bit on Asterisk and VoIP in general since I saw that some of your requirements
>> mentioned lines and intercom, 2 things that are kinda different when your in a VoIP
>> environment.
>>
>> What type of carrier technologies do you currently have for your landline (PRI,
>> analog, SIP trunk,...)? Also what type of link do you have in-between your offices
>> (mpls, vpn via internet, dedicated fiber,...) ?
>>
>> Thanks,
>>
>> ---
>> Andre Courchesne - Consultant
>> http://www.net-forces.com
>> MSN: courchea at net-forces.com
>> Skype: VoipForces
>>
>>
>>
>> L'information contenue dans le présent document est la propriété de Andre Courchesne. Et
>> est divulguée en toute confidentialité. Cette information ne doit pas être utilisée,
>> divulguée à d'autres personnes ou reproduite sans le consentement écrit explicite de
>> Andre Courchesne.
>>
>> The information contained in this document is confidential and property of Andre
>> Courchesne. It shall not be used, disclosed to others or reproduced without the express
>> written consent of Andre Courchesne.
>> Quoting "Bryan Welfel" <bryan at ashworthcreative.com>:
>>
>>> I am interested in hiring someone to design and implement a PBX phone system
>>> for our offices (currently across two locations). This includes configuring
>>> and training me on all software and telling me how to physically connect the
>>> system - I have an degree in information technology and am familiar with
>>> setting up servers / terminals. We have a server with Asterisk already
>>> installed and I will purchase 6 phones (recommendations are welcome). The
>>> job can be done entirely remotely (I can do any necessary hardware work) and
>>> the list of requirements are below:
>>>
>>> Requirements for PBX phone system:
>>>
>>> 1. When we receive an incoming call, we want the ability for the caller
>>> to select who they want to call and ONLY that person?s phone will ring. We
>>> will likely need to configure an automated voice prompt that lists employees
>>> to the caller. (ie someone calls and is greeted by voice recording that says
>>> press 1 for Bryan and press 2 for Joel. If the caller presses 1, Bryan?s
>>> phone is the only one in the office that will ring.)
>>> 1. In addition to requirement #1, even though a particular person?s
>>> phone rings, we still want to retain the ability for someone
>>> else to pick up
>>> the phone and take the incoming call.
>>> 2. We also want to have a backup if the person who the incoming caller
>>> is trying to reach doesn?t pick up his/her phone. There are two
>>> options if
>>> this happens:
>>> 1. Go directly to that person?s voicemail
>>> 2. Go back to main menu
>>> 2. In addition to requirement 1a, we also want to have multiple people
>>> jump in on the same line to engage in the same single conversation. (ie if
>>> Bryan is talking to a client on line 1, Joel and Joey can each pick up a
>>> phone and join Bryan?s conversation on line 1.)
>>> 3. If someone picks up the phone, they will have the ability to transfer
>>> the call to someone else in the office. Transfer requirements are as
>>> follows:
>>> 1. When someone transfers the call to someone else, the person?s phone
>>> that is receiving the transferred call will ring and will
>>> receive the call
>>> on the same line as the original call (ie if Chase wants to
>>> transfer a call
>>> on line 1 to Bryan, Bryan?s phone will ring and when he picks up
>>> the phone,
>>> he will pick up the call on line 1.)
>>> 1. If the person receiving the transferred call doesn?t pick up the
>>> phone and the ring limit is met, a default action will be taken. This
>>> default action can either be:
>>> 1. Go to the person?s voicemail
>>> 2. Go back to main menu
>>> 4. We want intercom functionality to function in exactly the same
>>> manner as the current phone system does now. That is, someone on intercom
>>> does not take up a calling line (ie Garrett want to talk to Isaac via
>>> intercom. He will be able to do this without taking up a line)
>>> 5. When someone picks up the phone, we don?t want the phone to
>>> automatically take up a line like the current phone system does. The phone
>>> only reserves a line when the person begins dialing a number or is receiving
>>> an incoming call.
>>> 6. We also desire the ability to push each person?s voicemail to their
>>> cell phone. (ie A client leaves a voicemail message for Bryan. The PBX
>>> system will push the voicemail to Bryan?s cell phone voicemail so he can
>>> listen and respond to it on the go.) (Note this is a bonus that is nice to
>>> have but can live without)
>>> 7. (Bonus) If the person who is the recipient of a call is not in the
>>> office, the call is transferred to their cell phone where he/she can answer
>>> it
>>>
>>> If you are interested in doing the work on site, we are located in
>>> Poughkeepsie, New York. Feel free to call or email me if you are interested.
>>>
>>> Thank you,
>>>
>>> Bryan Welfel
>>> 845.877.0410
>>> bryan at ashworthcreative.com
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> asterisk-dev mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-dev
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
More information about the asterisk-dev
mailing list