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Asterisk based Conferencing Solution

$250-750 USD

Cancelado
Publicado hace más de 14 años

$250-750 USD

Pagado a la entrega
We are looking for a consultant or a team of consultants to implement a teleconferencing solution for us. We would prefer Asterisk but if you have any other sugeestions ( sipXecs, OpenSIPS,etc.) we are open to discuss as long as the solution is proven and scalable. We need a Final solution which would: Allow us to hold multiple conferences simultaneously. We can start with 10 conferences but would need to expand to 50 or more in the future. Invite up to 10 people per conference Users should be able to join the conference by dialing a number through their phone or connect using their computer. Users could use a softphone but would need to be a web based softphone so no software installation would be necessary. Host\Users should be able to call-out to other participants (either on their landline or cell phone) and add them to the conference. Host\Users should have the facility to create a sub-conference while in a conference. The host\user should be able to mute\unmute all participants. The host\user would need a web interface to manage the conference This is a multi-phase project so we would need a solution which is scalable. The solution could be VOIP based but should be able to connect to users on PSTN lines. We would also need recommendations for hardware\software\VOIP providers for the Final Project once the project is awarded. Since this will be a multiphase project we are looking to establish a long term relation with the particular provider and the provider should be able to support the implementation in the future. Before we select a provider we would like to do an interview project. This posting is for the interview project. I repeat, this posting is for the interview project only and you are bidding for the interview project. Please don't bid if you don't have the team or expertise for the Final project. We will be selecting 3 providers for the interview project and depending on the deliverable of the interview project we will select the provider for the final project. We are looking for a long term relation so if you don't have the resource please don't bid. For the interview project we would like you to setup a test scenario where a user should be able to go a web page and join a conference, participate with other user and then dial out to another extension using the web dialer(phone). I know we will not be able dial out to landlines or cell phones but any other extension will also work. So for example lets say user 101, 102 , 103 & 104 are logged on to the system and registered with the server. User 101, 102 and 103 dial into conference 1234 and after a few mins of conversation 102 tries to dial in 104 for a one-on-one conversation with 101. So while 101 and 104 are having a private conversation 102 & 103 are still in the conference. The users should have a web interface to manage the conference and the ability to Mute/UnMute callers and have a private conversation with any person dialed into the conference. Please do not bid if your solution is to purchase an already existing design to be used (ie: WebEx, etc.). We must have this built from scratch to be completely customizable and scalable in all aspects. If you have done similar projects please provide references\links. We do not have test servers for the interview peoject so for test purposes everything should be self contained by you. The delivery time for the interview project will be 5-7 days.
ID del proyecto: 606840

Información sobre el proyecto

7 propuestas
Proyecto remoto
Activo hace 14 años

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7 freelancers están ofertando un promedio de $693 USD por este trabajo
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$700 USD en 7 días
4,8 (17 comentarios)
6,3
6,3
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Hello sir, if you check my reviews you will see that we have real experience developing and managing asterisk systems, and creating cms to manage users, calls, etc for asterisk. Please check your PMB for more info about me and my company. Regards, Pablo.-
$650 USD en 20 días
5,0 (2 comentarios)
3,0
3,0
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Dear Sir please check your pmb
$700 USD en 10 días
5,0 (1 comentario)
2,8
2,8
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please check PMB.
$750 USD en 10 días
0,0 (0 comentarios)
0,0
0,0
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I run a telephony infrastructure on Amazon's cloud (AWS). Based on this, scaling and redundancy are easily achieved, allowing for very efficient resource use (i.e., the available capacity scales transparently with demand, resulting in savings and efficiency). The infrastructure includes a conferencing service (based on "konference" product for Asterisk). I do have external PSTN SIP connectivity via SIP trunking and would be willing to do a full demo using real phones. In terms of web-based SIP endpoints, there is a variety - do you have any preferences for this? Regards, Igor Boguslavsky
$600 USD en 7 días
0,0 (0 comentarios)
0,0
0,0
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Details provided in PMB
$700 USD en 7 días
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Bandera de UNITED STATES
Mechanicsburg, United States
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Miembro desde feb 6, 2010

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