Voip Software Jobs

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Fixed-Price - Intermediate ($$) - Est. Budget: $50 - Posted
For my 2 Voip phones I have an account with Grasshopper. I need someone to comb through them and ensure incoming calls are routed to the correct v-mail. My struggle is configuring Google voice, Grasshopper & AT&T (cell) to work together. You should know going into this that I don't typically use data or WiFi and that seems to complicate matters for me. You should also know that I'm guessing, unless you are a super braniac that this project might take several hours. I need help simply because I can't keep track of all the different variables. variables being 6 different phone numbers for 2 people. I am open to ideas about how to smooth things out. Below are notes I took from when I tried to do this myself: My numbers are: 406-660-1828 - owned by Grasshopper, Voip 406-660-3063 - owned by Grasshopper, Voip 406-660-1075 - owned by AT&T, cell 406-660-0989 - owned by AT&T, cell 406-200-8367 - Google Voice 406-272-6520 - Google Voice 1828 to 1075 w/0 predial at 10:24am = 406-200-8367 on cell caller ID & my new voicemail msg. Get v-mail notification on cell phone. w/predial at 10:28am = 406-660-3064 on cell caller ID & my new voicemail msg. Get v-mail notification on cell phone. 1075 to 1828 at 10:31 = goes straight to new v-mail. Got Grasshopper notification in gmail along with transcript. Shows as ext 0 in Grasshopper. House phone to cell w/o predial = 272-6520 on ID & my new v-mail w/predial = 660-0989 on ID & my new v-mail (should be Starr’s) 1075 to 3063 = recording “please leave a message”. V-mail goes to Grasshopper as ext2 with no transcript in email. 1075 to 272-6520 = Starr’s voicemail Needs to be Where are my text messages? Can I segregate phone activity for my 2 phones? Need to change caller ID thingy for office phone to 1828. I shouldn’t have to do the caller ID thingy because my number is 1828. Can I get TM transcripts of my voicemails? So I can make the house phone look like 1075 by calling what number? Calls from office 1828 but show 1075. Then 1075 gets routed to 1828 when I’m home. So if 1075 is my main number. Calls go to 1075 AT&T Calls go to 1075 routed to Grasshopper. Can I get a TM notification? Calls from house 3063 but show 0989. Then 0989 gets routed to 3063 when Starr’s home.
Skills: VOIP Software
Hourly - Entry Level ($) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
We have a fresh FusionPBX box that is already working without any problem. Now we need to make some extra configs and we will need to complete next tasks. You must be very expert in FusionPBX / FreeSwitch. * Add native G729 protocol * Get a document to change / reset admin after moving / cloning VPS server to new IP * Implement mp3 natively to send * Install spanish and portuguese IVR / voices * Setup a couple of outgoing dialplans and configure default prefixes in 3 countries * Help configure tenants interconnection - we want to offer one single number accessible to all tenants * Be available for future configs / needs * We're interested in long term business relation You bid price per hour and the time you need to implement this (2h for an expert should more than enough) Thanks
Skills: VOIP Software Asterisk Freeswitch
Hourly - Entry Level ($) - Est. Time: More than 6 months, Less than 10 hrs/week - Posted
I am looking for somebody experienced with Twilio TwiML/API/Etc. who can setup a TwiML Bin for one of my numbers. I want 2 code examples to do the following. 1) Use any Caller ID of my choosing for call forwarding, as well as outbound calls. 2) Record both sides of the calls. 3) Use Twilio backend system to track call recordings per phone number. I want to be able to view ALL recordings for inbound/outbound calls. I will want to verify everything is setup correctly. (If you have a better suggestion I am all ears). I doubt this will take somebody very long, so I have no problem paying for an hour or so to have you walk me through the Twilio Admin System. I want this done ASAP, since I plan on launching my latest campaign in a day or so. I got the Call forwarding to work perfectly, but I cannot figure out the call recording part. Ideally, I would prefer to hire somebody on an ongoing basis to help me setup multiple campaigns with Twilio. I have a bunch of ideas I want to bounce of an experienced coder.
Skills: VOIP Software Twilio API VOIP Administration
Fixed-Price - Expert ($$$) - Est. Budget: $100 - Posted
Please ignore our budget and place your best bid for the project. We search for a expert to implement a VoIP SIP validation of a list of phone numbers. As a result of the validation we need as much as possible details about the given number. The goal is to reduce the manual effort of a call center by knowing if the given phone number is a valid or invalid one. Also it would be nice to know what type of availability/unavailability was the result of the phone validation. As a input you will get a CSV list. Your implementation shall be in Apache POI 3.14 and the column containing the phone number shall be configurable by a the column header name. We do not need a database, since we have a custom implementation. BUT your model classes have to properly designed to be later able to use it easily with a custom persistence layer. To ensure you are not a simple poster please add the result of eight power two on top of your application. As a result we expect to get a updated CSV file with the new columns which you need to provide for the end user to understand the status of the phone number validation. Ensure that your implementation has a component for CSV reading, one for VoIP validation and one for CSV writing. Ensure the components are independently usable! We prefer for VoIP / SIP Java Standards. As a runtime environment we expect runable on: - in JavaEE 7 (wildfly 10) - Java 8 As the development environment we expect: - Eclipse Neon - maven With your application provide us following details: - delivery date for milestone 1 - brief class level design with public methods - a list in word or excel of statuses collected for the later VoIP validation (see also https://de.wikipedia.org/wiki/SIP-Status-Codes) - delivery date for milestone 2 - working VoIP calls, no extraction of validation results - delivery date for milestone 3 - working validation extraction and implemented CSV writing
Skills: VOIP Software Apache Jakarta POI SIP
Fixed-Price - Intermediate ($$) - Est. Budget: $3,700 - Posted
Looking for PHP developer who have excellent experience in developing requirement based web portal having monitoring, reporting and exceptional GUI. For this project we are looking for someone who can develop a web based software integrating the features of the following existing softwares: 1) ViciDial 2) Go Auto Dial 3) DialFire 4) Avatar 5) Twilio etc Interested people please contact asap as we have an urgent requirement and let us know a time frame and budget for the same.
Skills: VOIP Software Asterisk FreePBX MySQL Administration
Hourly - Expert ($$$) - Est. Time: Less than 1 week, 10-30 hrs/week - Posted
Hi, Note: It is required that you have prior experience with provisioning Polycom UC phones using a provisioning server, please do not apply for this job if you do not have prior experience. Jo descriptions: We have a couple of Polycom VVX 500 and CX3000 UC Phones (Skype for Business enterprise voice). Requirement is to setup a Provisioning Server with required configuration files to automatically provision these Polycom phones. We already have following configuration completed: (1) Existing Skype for Business Server with DNS records for auto discovery, SSL etc. (2) FTP Server to use as a provisioning server (3) DHCP options added for Polycom phones To do: (1) Configure the master configuration file and phone specific files. (2) Customise Polycom configuration files for correct dial-plan and user specific settings for 6 Polycom VVX phones. (3) Customise Polycom configuration files for correct dial-plan and settings for CX3000. Desired outcome: (1) A new Polycom phone is plugged into the local area network. During the startup process phone automatically finds its configuration server using DHCP option. Phone configures itself using it's custom configuration file. Phones upgrades its firmware to the latest SfB compatible version. Phone is ready for use. (2) User makes changes to phone settings using interface in the phone. Every 1 hours, phone downloads it's custom configuration file from the provisioning server and reapplies it's custom settings, thus overriding user's settings. To know kind of work to be carried out, please check http://blog.schertz.name/2013/05/provisioning-polycom-sip-phones/ Estimated time: 16 hours (but you suggest how soon you can accomplish this)
Skills: VOIP Software Microsoft Lync Server SIP VOIP Administration