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    17,572 asterisk openser sip jobs found, pricing in USD

    We are a company with 15 sites in Portugal. with over 40 employees spread across the various centers. We intend to have a solution to talk between centers via IP, conference calls, IVR messages, automatic call forwarding, etc.

    $567 (Avg Bid)
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    Hello, i am working on a project. I have ordered a VOIP number, i want to automatically accept each incoming call to the number and record the call. I also need to get the recorded file via an api. Currently i don't know how to do this. I like to have an application that can be run either be run on windows, raspberry, in the cloud (i prefer cloud) Details about the phone number, sip number...

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    The project involves migration of existing Asterisk PBX to a Azure cloud. There is existing Azure account. Install FreePBX on Azure cloud. Install Linux and FreePBX GUI. After the installation, migrate exiting asterisk configuration and voice prompts and confirm that the system works accordingly. Copies of the Asterisk files will be provided (have to be specified). I would also like consultation...

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    Hi im looking for someone who will take asterisk config backup from existing centos 7 and then making installer for centos 7

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    Goautodial v4 Voip dialer crm experienced freelancer need for install and make SIP/Agent settings

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    Candidate should be familiar and comfortable with - Preference would be given to having experience on IVR and Call Center application development. - Hands on experience in Kamailio SIP Server - Integration with Database (PostgreS-Q-L) with knowledge of queries and stored procedures. - Experience in installing Hosted VoIP Candidate Profile The candidate must have: - 2 - 4 years of experience - Stro...

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    Hello. I'm look for someone to draw a set of plans for a log home with a day or two. 25' x 41' (center line) building. 8' basement. 8' first level with loft. 12" logs. 4/12 roof pitch. SIP 1st level floor and roof. East/West eve overhang - 4'. Gable end eves - 6'. I've attached The Trapper model home as an example to what I'm looking for....

    $50 / hr (Avg Bid)
    Featured Urgent
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    We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass recorded audio as voice calls over SIP and forward them through whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows whatsapp executables, or by using the Android / Windows Phone mobile ve...

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    Details will be shared winning bidder. please bid if you have the experience.

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    Call transfers on PBX 4 days left
    VERIFIED

    To make assisted transfers on FreePBX/Asterisk with one click. FreePBX / Asterisk with WebRTC ! This will be the first milestone to be discussed. For the whole projrct, see attachment.

    $426 (Avg Bid)
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    VICIDIAL Customization -- 2 3 days left
    VERIFIED

    Need a call center solution using VICIDIAL, Need all the standard features of Vicidial and it should work with Asterisk 1.6 or 1.8 or both. (or higher) All Vicidial UI and branding has to be replaced with our custom brand name, and interface should be changes sufficiently to look different. Logo branding removed Marketing automation Lead analytics dashboard Analytics dashboards SMS +++ To Email ...

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    i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk w...

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    astreisk_rules 2 days left

    i want make rules in my server 1- i want port number 1 call number 2 to i mint 2 i want when i send call for port number one when it finsh it make stop this port 2 or 3 min exe exten => _[*#0-9]!,n,Dial(Mobile/A0/${EXTEN},60,tTm,K) exten => _[*#0-9]!,n,Dial(Mobile/A1/${EXTEN},60,tTm,K) --------------- A0 port number 1 A1 port number 2 If a user make a call from a sip account, asterisk w...

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    We need an app developed on kazoo monster-ui so that SIP Phones/Devices can be auto provisioned just like this: [login to view URL] The script in the video is discontinued/not working as it was for version kazoo 3.x, and the latest kazoo version is 4.3 (major update). I don't need every handset brand/model, just 1 model but the app should be flexible so that we can add more brands/models in...

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    WebRTC-Asterisk-FreePBX 1 day left
    VERIFIED

    PLEASE READ THIS FIRST ! Asterisk with WebRTC for Inbound call center. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. There are a few fixes to make (see attachment). ...

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    High level description: The IVR need to "collect" digits from the coller (personal ID number) via DTMF or Google voice to text service. The collected info needs to be stored on SQL table and passed as a parameter to dynamic (user defined) URL Task prerequisite: Asterisk 11, freepbx 2.11 Asterisk 16, freepbx 15 Full installation doc. All parameters will be via php page on /var/www...

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    Develop ionic application where SIP calls can be made

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    VoIP Cloud PBX migration 19 hours left
    VERIFIED

    We are seeking a VoIP Engineer to move clients from Kerio Operator to Vital PBX You will need to configure Vital PBX with the same setup/settings the client current has in Kerio Operator. No client is to go live on the new PBX, this is just setup without activating the SIP Account. First test will be our office to go live, pending no issues one by one we will do the final step to turn other cli...

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    Build phone system I am looking to set up sip trunking/pbx etc. I'm wondering if you are able to help me. I'm not very knowledgeable on the subject matter but i'll tell you what I have, and what I need. I have: - a VPS server with Centos 7 installed - a Twilio account with 2 numbers -number 1 is being forwarded directly from Twilio to my cell phone -number 2 is involving...

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    Hi, we need an experienced Xamarin developer to develop a cross-platform (Android/iOS) chatting app for internal use of a company. The project is split into phases. The first phase is a VoIP SIP softphone that supports Voice calling, Instant Messaging, File Transfer and Presence. Exactly the same as Whatsapp. The following features are required in the app for this phase of the project: - Voice ...

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    sip asterisk specialist to get in touch with our team in order to drive inner development of a core product. SIP webrtc, SIP API sip asterisk master configuration

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    We need an expert in jack audio connection tool box to create virtual cable between sip client and android speaker and mic

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    PLEASE READ THIS FIRST ! Asterisk with WebRTC - A real Specialist is needed.! For Inbound call center with Asterisk and WebRTC. This is only for freelancers with a very broad knowledge-base and who know how to get things done with the terminal. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubun...

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    I have mostly setup the Twilio account already but need here/there help to understand the thing fully and be able to continue developing. I would need you for 1-2h per day and probably need 5-10 days in total. I will do homework between days. We'll be using Chrome remote desktop, TeamViewer, [login to view URL] or whatever so you see my screen and can instruct me. First tasks Get Twilio work...

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    Hi Ghanshyam P., I noticed your profile and would like to offer you my project. I am looking for sip dialer customization. Let me know if you are interested.

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    Hi Sunil C., I noticed your profile and would like to offer you my project. I am looking for SIP Android and IoS app developer. Please let me know if you are interested.

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    Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and mos...

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    Hello, I am looking for a data scraper who can get me leads from America from Instagram for the personal trainer niche. I need 1,000 leads for now. If you do a good job, I will hire you for a lot more business. I need the following from each personal trainer / health coach’s Instagram: First name*, last name, business name, phone number* email address*, City*, State*. Anything in the asteri...

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    Asterisk with WebRTC Specialist needed.! PLEASE READ THIS FIRST ! For Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical Servers are installed and mo...

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    An application is required where mainly SIP calls can be made. It also requires that the development platform be hybridized, specifically in Ionic.

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    Hi Mohammad Abu S., I contacted you on fivvr. I never knew you also existed on Freelancer. You follow the rules well. I wonder if you would be open to helping me fix my freepbx installation. The only trouble I have is: (1) Receiving calls in from twilio sip trunk. It connects but every now and then it doesn't. Looking into the logs, it seams to be because twilio sends from 4 different ...

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    Build React Native VoIP App/SDK with SIP protocol which able to integrate with current React Native solution. Feature/Function that require as below: • Push Notifications • Receive VoIP calls on the lock screen, active VoIP calls are no longer disrupted by incoming cellular calls and VoIP calls are now stored in the native applications call log. • Call Transfer • Call Record...

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    I want simple dialpad with all dial functionality in web application (html as ui) and call flow should be initiated from asterisk / pbx

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    I need someone to set up an authenticated SIP trunk on an audio codes SBC that is then connected to a Lync server. Lync servers do not allow authenticated trunks and as such the SBC is required to do the authentication with the external SIP server then pass it on to the Lync server. I am looking for someone who has already done this before. Not someone that just has experience in these technologie...

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    We require a program which will change and manipulate SIP parameters in realtime to enable us to test end to end equipment response.

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    Fixing and writing new asterisk codes for our servers.

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    we need a expert in Linux that will create a very small Linux distro with 3 Main functions 1. BlueZ (virtual Bluetooth server) [login to view URL] 2. config [login to view URL] virtual audio cables 3. Linphone open source VoIP SIP softphone the small distro when it run will receive audio from a Bluetooth device that was pointed at the BlueZ and with the virtual audio cable will be able to trans...

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    We need an expert in Linux to create a very small Linux Distribution. Could act as virtual bluetooth [login to view URL] Linux BlueZ server (virtual Bluetooth server) we need a expert in Linux that will create a very small Linux distro with 3 Main functions 1. BlueZ (virtual Bluetooth server) [login to view URL] 2. config [login to view URL] virtual audio cables 3. Linphone open source V...

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    we need a expert in Linux that will create a very small Linux distro with 3 Main functions 1. BlueZ (virtual Bluetooth server) [login to view URL] 2. config [login to view URL] virtual audio cables 3. Linphone open source VoIP SIP softphone the small distro when it run will receive audio from a Bluetooth device that was pointed at the BlueZ and with the virtual audio cable ...

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    Our company is in the process of moving many of our services. We would like to have a Asterisk VOIP Server installed and configured on a new AWS Instance under our AWS account. Candidate must have experience with Asterisk. [login to view URL]

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    Hi! looking for somebody who can help me in developing Financial Calculators for my financial advisory business...

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    I need a design of a table sign for Chicago bars and restaurants who want to reduce plastic straw waste and provide plastic straws only upon request (non automatically) or switched to naturally compostable/reusable straws upon request. I have attached two "drafts" and our logo. I'd like you to make something beautiful and attractive. The main message: skip the plastic straw secon...

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    Looking for a developer with experience working with Asterisk and FreePBX. Looking for a billing solution.

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    We need a freelancer for full Asterisk configuration and implementation in our organization with complete Trunk Configuration(GSM Gateway) and free complete Mobile application customized for our Organization.

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    New enhancements on asterisk dialplan I have 2 termination partner. Company A and B. They have their own DIDs to mask. They will provide their DIDs and we will store them in dids table to mask original calling ID. Call will come asterisk will choose one DID from did table and send the call to termination partner. it can be "company a" or "company b" but if it chooses comp...

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    We will provide you with a clean install of Alpine Linux running on a raspberry pi. You install and configure the following software. hylafax+, iaxmodem, asterisk, freepbx We will provide you with port 22 ssh access to this device, we can also provide additional access if you require it.

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    Need experienced individual to set up SIP in combination with VOIP serice utilizing Virtual SIP phone.

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    I am looking for someone who can build a sip calling APP using a Mobile Hybrid platform such as ionic or react native or any other Mobile Hybrid platform. the requirement is to connect to our SIP servers and make calls through an outbound SIP proxy. we will provide all the server details needed user needs to enter username, password, domain, proxy and outbound proxy, user needs to register using ...

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    JobDiva VoIP Integration Project Details, 1- Click to Dial: JobDiva supports Click to Dial using HTTP call or RESTFul API. Please check with the VOIP provider for the method that is supported and let us know the details. 2- Incoming Call Screen Pop: two methods are supported. a. Use the JobDiva API web services’ searchCandidateProfile and/or searchContact method to locate a matching can...

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    PLEASE READ THIS FIRST ! -- Inbound call center with Asterisk and WebRTC for agent access. This is only for freelancers with a very broad knowledge-base. Some requirements: FreePBX/Asterisk, Java Script, pjSip, Bootstrap, CSS, HTML5, PHP, Web Dev., Stun / Turn Servers, Queues / Parking for calls, SSL, Ubuntu, etc. The physical setup is done and most of the software is installed. There are a few...

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