VoIP Jobs and Contests

Voice over Internet Protocol or VoIP refers to a host of transmission technologies meant to deliver voice communications over IP networks that include packet switched networks and Internet. VoIP systems include communication services such as facsimile, voice, and voice messaging applications. A number of firms are shifting to this system to promote a cost effective medium of communication. You can hire an expert who can help your firm to leverage the potential of this new system. Just post a job today!
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Browse Jobs on Freelancer

Project/Contest Description Bids/Entries Skills Started Ends Price (USD)
Write some Software We are looking for a developer to build a simple voice mail application using .NET libraries. Specs will be provided to qualified developers. The voicemail will need to receive calls from internal extensions and external callers and play a prompts. After playing the prompt it will record the callers message. 9 .NET, VoIP, VoiceXML Jun 22, 2017 Today6d 15h $610
Complete VOIP solution for GoIP Equipment The main part of a project is a creation of software which will help to manage GoIP equipment (Hybertone manufacturer - gateways and sim banks). It should be probably based on a softswitch (Asterisk is NOT OK here). All the features which will be included should be available in the interface - Some basic features are: 1:Simulation of Human behavior -Generation of the flow of incoming calls -The list of 'preferred' numbers. -Imitation of SIM card movement around the city. -Generation of the flow of incoming SMS and USSD requests. -Implementation of daily and weekly cycles of human activity. -Time slots for activation and time slots between calls. 2: SIM cards Operations -The IMEI substitution using an actual base of devices. -Black lists and white lists of phone numbers to protect from the operator’s calls. -Automated tracking of SIM cards operation to prevent blocking. -Simulation of human behavior on GSM networks. 3:Automation -Automatic activation and loading of SIM-cards at the pre-set time. -Automatic connection of bonuses. -Sending and processing of SMS / USSD-requests to check the balance. -Creating patterns to top up when the account predetermined amount is spent. -Checking and unloading of the blocked SIM-card. 4:Statistics and Monitoring -Counter of the number and duration of calls. -Counters to reset at the specified time. -Live statistical data on the performance of your SIM-cards. -ASR and ACD for each SIM-card and channel -Channel performance status. -RSSI (signal strength) of each channel. -The duration of the current call. -Activity and statistics for each SIM-card. -Rapid notification of problems via Skype, SMS or email. 5: Built-in SIP server to work with SIM card bank. We can show you the cases of the similar services which work with GoIP equipment so you can get better idea. 30 PHP, Software Architecture, VoIP, Telecommunications Engineering Jun 21, 2017 Jun 21, 20175d 18h $2514
EXPERT VICI-DIAL PROVIDER MUST HAVE OWN VICI-DIALER FOR CALLCENTER 5 -- 2 EXPERT VICI-DIAL PROVIDER MUST HAVE OWN VICI-DIALER FOR CALLCENTER MUST HAVE OWN VICI-DIALER ! MUST HAVE OWN VICI-DIALER ! - looking for vici dial provider with own vici dialer to service call center We are now paying 1,4 cents a minute all included calling the usa We are looking for a better rate Tell me your best rate per minute Also tell us why we should pick you! 5 Telemarketing, Asterisk PBX, VoIP, MySQL, FreeSwitch Jun 21, 2017 Jun 21, 20175d 12h $11
configure vici dial server Configuration of vici dial is required, engineer must know how to configure ip based and user based VOIP account. This can be both vici dial & goautodial as its a ongoing project. 4 Apache, VoIP, Windows Server Jun 21, 2017 Jun 21, 20175d 4h $130
sip dialer and server I am looking for a freelancer to help me with my project. The skills required are Asterisk PBX, Cisco, FreeSwitch and VoIP. I am happy to pay a fixed priced and my budget is $250 - $750 USD. I have not provided a detailed description and have not uploaded any files. 12 Asterisk PBX, Cisco, VoIP, FreeSwitch Jun 20, 2017 Jun 20, 20174d 17h $590
Project for Asterisker Hi Asterisker, I noticed your profile and would like to offer you my project. We can discuss any details over chat. In fact we want to customize LinPhone ([url removed, login to view]) in order to have 3 things: 1. Keep current features of Linphone 2. Customize User Interface so that users can see Call Bundles offer, purchase a bundle and user for calls 3. Create a Backend Admin Panel from where we can create call bundles with prices, check download, usage and call statistics per user, daily, weekly, monthly, etc. Manage users, receive their complaints and assist them. If that is a work you can handle, please ping me. 2 PHP, System Admin, Asterisk PBX, VoIP, , VPS Jun 20, 2017 Jun 20, 20177d 7h $12
طراحی اپلیکیشن تلگرام غیر رسیمی بنده یک سورس کامل از امکانات تلگرام همانند موبوگرام دارم فقط می خواهم یکسری امکانات مثل تماس صوتی و تصویری کاملا مجزا بر پایه سرویس webRTC به اپلیکیشن اضافه شود لطفا فقط افرادی که تجربه کار با api تلگرام را دارند پیشنهاد ارسال کنند 51 Java, Mobile Phone, Android, VoIP Jun 20, 2017 Jun 20, 20174d 3h $39460
EXPERT VICI-DIAL PROVIDER MUST HAVE OWN VICI-DIALER FOR CALLCENTER 5 EXPERT VICI-DIAL PROVIDER MUST HAVE OWN VICI-DIALER FOR CALLCENTER MUST HAVE OWN VICI-DIALER ! MUST HAVE OWN VICI-DIALER ! - looking for vici dial provider with own vici dialer to service call center We are now paying 1,4 cents a minute all included calling the usa We are looking for a better rate Tell me your best rate per minute Also tell us why we should pick you! 0 Telemarketing, Asterisk PBX, VoIP, MySQL, FreeSwitch Jun 19, 2017 Jun 19, 20173d 15h -
FreeSwitch/Fusion PBX SIP Gateway Expert Need We need help to configure custom SIP gateway in our FusionPBX for the outbound calling service. Currently our SIP provider unable to provide us direct SIP settings to configure them with our FusionPBX as they only providing us Mobile Dialer based SIP client which we are unable to configure as SIP gateway. We want to configure that Mobile SIP Dialer (SIP Settings) with our FusionPBX Gateway Profile/SIP Profile in order to add multiple gateways. We will share project details on the interview. This job was posted from mobile device, so please pardon any typos. 14 XML, Asterisk PBX, VoIP, FreeSwitch Jun 19, 2017 Jun 19, 20172d 23h $31
Virtual-alarm-receiver-ASTERISK I am looking for a freelancer to help me with my project. The skills required are Asterisk PBX, Cisco, FreeSwitch and VoIP. I am happy to pay a fixed priced and my budget is $250 - $750 USD. I have not provided a detailed description and have not uploaded any files. 10 Asterisk PBX, Cisco, VoIP, FreeSwitch Jun 18, 2017 Jun 18, 20172d 14h $606
Fusionpbx using webRTC and IP phone We have a working fusionpbx server that makes calls with webRTC extensions. We want to add support for IP phones so a physical phone can be used to make calls from the webRTC interface: * User can make a call from webRTC, and the IP phone will ring. * User can see an incoming call and answer it from the webRTC interface. More informations in private 2 Asterisk PBX, VoIP, FreeSwitch Jun 18, 2017 Jun 18, 20172d 8h $256
Asterisk Developer - Fixing and developing we are looking for a developer that expert at IVR systems based Asterisk we have our systems and we want to make changes, and developing at them. 13 PHP, Asterisk PBX, VoIP Jun 18, 2017 Jun 18, 20172d 8h $180
Aironet 1300 - open to bidding I have a wireless bridge using Cisco Aironet 1300. The throughput is only about 3 megabits per sec. Can you help to fix 5 Asterisk PBX, Cisco, VoIP, Network Administration, FreeSwitch Jun 18, 2017 Jun 18, 20172d $211
Aironet 1300 - open to bidding I have a wireless bridge using Cisco Aironet 1300. The throughput is only about 3 megabits per sec. Can you help to fix 6 Asterisk PBX, Cisco, VoIP, Network Administration, FreeSwitch Jun 18, 2017 Jun 18, 20171d 22h $161
assist with bulk sms I am looking for a freelancer to help me with my project. The skills required are SMS Bulk Marketing, Email Marketing, Interspire and VoIP. I am happy to pay a fixed priced and my budget is $250 - $750 USD. I have not provided a detailed description and have not uploaded any files. 7 Bulk Marketing, VoIP, Interspire, Email Marketing Jun 17, 2017 Jun 17, 20171d 16h $407
Cisco voice CUCM, UCCX, CUC CUCM, UCCX, CUC, gateways, CUBE 12 VoIP Jun 17, 2017 Jun 17, 20171d 14h $551
Configuring FreePBX system for Cisco USECALLMANAGER I need a working FreePBX installation with three extension set (cisco 7975G) and one single sip trunk. I will give you SSH and HTTP access to the Virtual machine with a fresh FreePBX Distro installed. FreePBX have to be the latest with asterisk 13 13.14.0 and the Usecallmanager patch for the same. I will need the [url removed, login to view] for the three cisco phones. 21 Linux, Asterisk PBX, Cisco, VoIP Jun 17, 2017 Jun 17, 20171d 9h $199
Asterisk APP with some API actions Hello, I need an Asterisk application that will do "call forwarding" with some API actions. Requirements: 1-) Adding new did numbers via HTTP API (like http://ip/[url removed, login to view]) 2-) Removing new did numbers via HTTP API (like http://ip/[url removed, login to view]) 3-) When someone called one of from our DID numbers, ask to our web service and forward the call to the number given by web service. Like: - When someone called, call [url removed, login to view] - Web service will answer you the phone number. - Forward the call to this number. 4-) Call status reporting for ANSWER/BUSY/UNAVAILABLE/HANUGUP/ERROR actions to our web service with duration & did numbers (our did and caller id). Thank you. 9 Asterisk PBX, VoIP Jun 16, 2017 Jun 16, 201721h 47m $203
Asterisk APP with some API actions Hello, I need an Asterisk application that will do "call forwarding" with some API actions. Requirements: 1-) Adding new did numbers via HTTP API (like http://ip/[url removed, login to view]) 2-) Removing new did numbers via HTTP API (like http://ip/[url removed, login to view]) 3-) When someone called one of from our DID numbers, ask to our web service and forward the call to the number given by web service. Like: - When someone called, call [url removed, login to view] - Web service will answer you the phone number. - Forward the call to this number. 4-) Call status reporting for ANSWER/BUSY/UNAVAILABLE/HANUGUP/ERROR actions to our web service with duration & did numbers (our did and caller id). Thank you. 4 Asterisk PBX, VoIP Jun 16, 2017 Jun 16, 201720h 43m $392
Spokesperson/Face of Empowerment for Women I am looking for a spokeswoman 19-25 years old to appear in small print campaign, website/social media, and speaking engagements for a start up company who wants to empower young girls and women to have a voice and be heard. Being a dog lover is a plus! 5 Video Services, Print, VoIP, Fashion Modeling, Social Media Marketing Jun 16, 2017 Jun 16, 201713h 36m $148
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