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Freepbx Jobs

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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
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: $5 - Posted
We will send calls to Asterix and we want to create those calls like brand new calls. and we will send them to provider. We want to manipulate all call details. All sip pockets like a new one. When we do this we want to be able to do the things below. A list of CLIs, or need to generate random CLIs based on our own rules, can we handle them. We need to map random or certain CLIs to incoming traffic so when your receive traffic, always the same CLI is assigned to the same destination. We need to regenerate your incoming calls and send them out like a new call generated from your network?
Skills: FreePBX Asterisk VOIP Administration VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $200 - Posted
We want to have Agent CallbackLogin Option (Not CallBack Extension). Is there any possiblity to implement that feature and how much it will cost for customization and time frame. (Basically What I need is: 1. Agent picks up the phone, dials 200 and listens to the agent message. 2. Enters the agent number,# and listens to a message asking about the extension (which doesn't happen in the above scenario). 3. After entering the extension,# it gets "permanently" logged in on that extension even after he hangs up. 4. He stays logged in until he logs off by doing the same procedure as above only without entering any extension number (agent number,#,#). I want to use elastix call center because of agent reporting and all Thanks and regards,
Skills: FreePBX Asterisk Elastix VOIP Software
Fixed-Price - Intermediate ($$) - Est. Budget: $150 - Posted
Installation and configuration Asterisk v13 FreePBX v13 and A2Billing v2.2 on a remote vps or Installation and configuration freeswitch fusionpbx with billing installation on the vps if you're available.
Skills: FreePBX A2Billing Asterisk CentOS
Fixed-Price - Intermediate ($$) - Est. Budget: $30 - Posted
Having issue with FreePBX configuaration not working properly. Extension forward kicks in voice mail immediately and disconnects the forwarded number i.e. extension 107 forwards to my cell phone number and after one ring, phone goes into voice mail and cell phone is disconnected.
Skills: FreePBX
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