Multitenat PBX based on asterisk / freeswitch  

your work, added work and compensation for future development and would be available.


Additional files submitted:
Wishlist.xls

Ds


Admin Panel
1.A control Panel to manage the customers .which includes
2.Assiging Credit to them
3.Seeing list of customer with numbers
4.Deleting a Customer
5.Ability to upload DIDs .
6.Ability to see the calls in progress .
7.Ability to activate / deactive a DID or DIDs.
8.Ability to add a country and than upload DIDs / Assign DIDs.
9 Ability to add CLI restriction.




regards,
alamzaib

and would like to be able to create a small desktop application which users of the pbx can use to call contacts that are store on contact folders in outlook or exchange public contact folders.
Each call will be logged in an ms sql, firebird, mysql or access database and reports can be created against the calls.

Thanks in advance

Thomas

*Welcome message* enter your account number
-user dials account number
System: Enter your password
-user dials password
System: To check in press 1, to check out press 2
-user Dials 1 to check in

I need asterisk expert or freeswitch to work on system to be configured to route and handle the incoming calls based on defined parameters.

the customer Database
If it is, then Jane's caller ID is changed to 212-888-1212
If not then Jane's caller ID stays 310-555-1212

We can run a dump from the customer database to create a csv file that has Customer phone number, Store Phone number. We'll update the csv file nightly. But it will always be in the exact same location on the asterisk server and the format will never change.

l setup. Although, please mention an estimate of the complete setup in a PM.

The right candidate should have multiple experiences setting up a similar cluster. If possible should be able to show us the work and provide references of current or past customers for whom you did a similar setup.

The first screen of the script will ask the users to enter the extension he is logged in as.
The script will then monitor AMI for the call ringing in that extension and accordingly capture the DNID value for that ring and push open a new browser and ope the url.

I hope I have been able to explain this well.
If there are any questions, please feel free to PM.

Thx
Sans

ule.
currently using redirect 300 302 but this requires additional setup on client gateway
need to use t_relay or forward but be able to authenticate via source IP.
Prefer custom sip header, but we are open to suggestion.
need user/name authentication as well as IP authentication preferably on asterisk but it can be perform on openser without alteration of a2billing database structure.

entire Asterisk server, and helping us find all the respective service providers we need to make our business a reality.

BUDGET: I was unsure what to put for the budget, so please disregard the $ 3,000 - $ 5,000 budget and bid what you truly believe the cost would be. We have a large enough budget, but would like you to place your bid on what you believe it will cost to execute this system.

i want to use in my Asterisk server i have an 'one way audio' problem , in freeswitch some kinda of 'unknown media type' and stuff. So, i can shoot that are problem with my server.
By the Way i need specifcly that´s my server works correct with SRTP-SDES and TLS-SIPS . I have in my server Asterisk and freesitch already Instaled.
In first moment as related above is just an server correct ajust.

Will Be Waiting The Bids


Mario-Brazil

when i want to use in my Asterisk server i have an 'one way audio' problem , in freeswitch some kinda of 'unknown media type' and stuff. So, i can shoot that are problem with my server.
By the Way i need specifcly that´s my server works correct with SRTP-SDES and TLS-SIPS . I have in my server Asterisk and freesitch already Instaled.
In first moment as related above is just an server correct ajust.

Will Be Waiting The Bids


Mario-Brazil

system will call on eyebeam and agent need to attend the call. If agent attends the call, agent will login.
* Can answer incoming calls
* Can hangup calls
* and Iframe area: call an url with caller id. For example; if 901221112233 id is calling us; display http://xx.com/yy.php?caller=901221112233 in an Iframe. I will give informations about customer at here.

No outband features I want. Only these are.

Thank you.

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Additional files submitted: (Files are only available for logged in users)
CallFlowDesign_.pdf

I require a remote install of a2billing on asterisk server, files have been downloaded and untarred ready for install. You must have knowledge and experience in both areas as incoreect install can damage Asterisk

We want to setup a simple Voice Broadcasting System based on Asterisk / Elastix. Teleyapper is a good app for voice broadcasting based on asterisk but we need a GUI for settings and execution.

Support Click to call functionality
7) Should support codec conversion
8) should work in Nimbuzz & Fring
9) LCR Functionality
10) should support video calling
11) should support multi-lingual IVR
12) Should play a greetings before a call is connected if called by dialing 00
Please let me know whether you can provide a switch with all the above features. Please feel free to contact me for any further clarification.

Best Regards,

sed on the areacode being dialed
there will be at least 5 groups

All other Extnsions need to work mormally

in a nutshell
if extensions 700 to 799 call 91678NNNNNNN ,callerid=7705550121
if extensions 700 to 799 call 91770NNNNNNN ,callerid=7705550121
if extensions 700 to 799 call 91706NNNNNNN ,callerid=7061110121

Caller ID is set at the extension level, for all the other extensions. 100-699, this needs to continue to work.

valuable time and interest.


Additional files submitted:
AsteriskEN.jpg

supply graphics and front page web pages so work is mostly focused on admin panels and actual telecom service.

We require a full and professional analysis of project and clear delivery milestones for payment plan

We only would consider working with people with proven record/demos. Please provide links and access codes in PM
serious bidders only with good knowledge of Asterisk A2billing, IVR, DID routing, PHP.

More details in PM

I have successfully installed Asterisk based Application Astbill (http://astbill.com) on my linux box. What I need is someone to help to create Provider Trunk and Route from different providers so that user can use cheapest calling rate avalaible to make call to their destination. You will need to know ASterisk and Astbill in order to help.