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Freepbx Jobs

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Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
Developer will prepare server, install and configure A2Billing and relevant components. Absolutely required features: - Callback - web callback; Ani callback - Calling card - Call shop - Reseller agent accounts - Trucking - Origination and termination - Asterisk PBS programming - Custom IVR upload and configuration - In addition to standard features. There is a potential for service support on an ongoing basis upon successful completion of project. Developer must be flexible and punctual.
Skills: FreePBX Asterisk HTML5 Linux System Administration
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
Hello, This is a Example Description Description Kdeals-app is a free VoIP audio and video Dialer to make VoIP call from the Android Phones,Iphone-Phones,Blackberry,window/Nokia & Tablets. It works over Wi-Fi/2G/3G/4G/EDGE/HSDPA kdeals Dialer Must support high quality audio & video codecs like, G729,G729a, G.726 ,G711a, G711u, g722, VP8, H.263-1998, H.264 etc. Compatible with any SIP based Softwitch Connectivity through WIFI/3G/4G/Edge/HSDPA Login by Your SIP server, SIP user & Password Outgoing and incoming VoIP calls Address book integration Calls History Loudspeaker Microphone on / off work in background mode Excellent Voice Quality Audio Codecs Support, G729, G711a, G711u, G722, AMR & GSM,PCMA/G.711 A-law,PCMU/G.u-law Video Codecs Support, H.263, H.264, VP8 AMR Speex16khz Sppex8khz Speex 32 Khz Opus 48khz Silk 24khz Silk 16khz iLBC AAC-ELD 22khz AAC-ELD 44khz Compatible with Android Phones & Tablets OS version 2.6 and above ETC....... Codec Section for Voice: Codec Section: Speex 32 kHz >> Automatic Selected Speex16khz >> automatic Selected Speex8khz Opus 48khz Silk 8khz Silk 12khz Silk 16khz Silk 24khz >> automatic Selected GSM 8 kHz AAC-ELD 22khz AAC-ELD 44khz ISAC 16 kHz ILBC 8 kHz AMR 8 kHz AMR-WB 16 kHz CODE2 8 kHz G729a >> automatic Selected G729 >> automatic Selected G711a G711u G722 G726-32 8 kHz G726-24 8 kHz G726-16 8 kHz PCMU 8 kHz >> automatic Selected PUCMA 8 kH8 >> automatic Selected RFC-2833 >> Automatic Selected QUAD-BAND >>GSM850, automatic Selected GSM900, >> automatic Selected GSM1800 ,>> automatic Selected GSM1900 >> Automatica Selected Support: Description : This app must be a customize SIP/XMPP based VOIP/IM app as my requirement and have ready for use without no Delay Registration when Consumers/Customers download the application.In all Mobile-voip-app Registration is depending on the Developer with code implementation during the production of the application You can review the following above in ( Description ) which will be needed from you to make the application running as a app in perfect condition. The application must work on open source Android/iPhone/web/windows mobile 8/mac/blackberry/Symbian SIP apps those work with Sip server like asterisk. For Example the app must also work on asterisk AMI (Asterisk management interface),AGI (Asterisk gateway interface). 2) Below Must be integrated as a Default when application download, See Below: Registration expire time >> 3600 Transport type >> UDP=5060 to 5080 Stunt Refresh Period == 30000 Send RTP DTMF Send SIP DTMF Send SIP INFO DTMF Noise suppression 2) Under >> Network, See Below: Registration expire time >> 3600 Transport type >> UDP=5060 to 5080 and UDP 10000 to 30000) RFC-2833 Stunt Refresh Period == 30000SIP INFO == automatically Selected 2+) We will Explain you about Complete network. 3) I already Design the presentation of the app how it should presented on a mobile-phone etc.. when downloaded by consumers, So in this case you would only need to Create your apk and implement the functionality of the application, in this email attachment you will see a picture which should be in the application which will come under Codec. ( Tab), when consumers click on codec's which should give them a option to Choose the codec which they wish to use during they Call. and the other image/picture in the attachment Must come under ( settings ) which give the consumer option to Configure the application for use of Voip-calls Chat-flow 4)Regarding My VOIP/ SIP app, as I discussed/mention the implementation of my requirements for chat-plow. Please check features list and let me know, I if you understand: - Personal Chat/ Group Chat - Send/ Receive Pictures/ Audio/ Video files through app - Conference audio/ video calls. Video call will be upto 4 persons at a time - Send / Receive emails similar to BlackBerry Messenger app. User can configure unlimited email accounts in the app and send/ receive email through - Offline message, user can send SMS from itself but received SMS will be saved into the native massage app of the device. Will use messages APIs to send unlimited messages. -Free Calling within the app. - Sending and receiving picture within app. - SIP support: app user can make call on mobile and landline numbers by getting services from SIP providers. Please get back to me with your latest feedback and do let me know if you understand the features list of the app. Look forward to hear . 5) No Mile-stones are no Escrow will be accepted because 99.9% of developers are not Honest they are only/telling lies,when the job is completed and applications is tested the money will be Paid to the Developer Through ELANCE. Thank you Regards Rakesh Singh
Skills: FreePBX Adobe Photoshop Android App Development Asterisk
Fixed-Price - Intermediate ($$) - Est. Budget: $300 - Posted
We need someone very familiar with the setup, customization, administration, and usage of Vicidial software. We need the following: -Vicidial configured to work with our Asterisk system and / or with our SIP carrier -5 Agent profiles created completely and ready for use in campaigns -Template campaign setup that we can easily copy and change the contact list -Any customer agent interface that you have that would be nicer than the default -Ongoing administration and support
Skills: FreePBX Asterisk vicidial
Hourly - Intermediate ($$) - Est. Time: Less than 1 week, Less than 10 hrs/week - Posted
A) On a Raspberry Pi 2, configuration of RasPBX (FreePBX on a Raspberry Pi) for a small office environment with the following: - 6 PSTN lines (trunks) One main telephone number cascading - 1 PSTN fax line, dedicated - A GXW-4108 Grandstream VoIP gateway (experience in configuration of this gateway is required) - 10 extensions - IVR attendant to direct calls to appropriate extension or cell if extension not answering - Possibly will also require 1 GSM gateway B) The job is required for March 10th 2016 but I will need a quote asap.
Skills: FreePBX Asterisk Raspberry Pi
Fixed-Price - Intermediate ($$) - Est. Budget: $1,000 - Posted
Hello We want to develop and automated telephone system we have an asterisk server which is not operational yet. We have an SQL server table where Vehicle and its pin code is saved. What we need is? When an entry comes to an SQL Server table lets say alarm (I can copy that to MYSQL or Asterisk Database) then aasterisk should automatically calls the number and ask "PLEASE ENTER PIN CODE" as user enters it would authenticates the PIN and then Ask the USER"Are you travelling in vehicle?" If he press one then No issues if press 2 then logs then entry in the database from where I will copy it to SQL SERVER BACK.
Skills: FreePBX Asterisk SIP VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $50 - Posted
We are currently unable to save/pull CDR information from FreePBX 12 because the system was custom setup instead of from the Distro ISO.
Skills: FreePBX Asterisk
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