Need develop simple sip api allow create product jobs

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    10,297 need develop simple sip api allow create product jobs found, pricing in USD
    Allow List 5 days left

    I am looking for a mobile app developer that can get their work done quickly and efficiently, while communicating effectively. The app will need a simple UI, with ability to work withing the call and contact features of the phone. More info will be given to selected dev.

    $190 (Avg Bid)
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    12 bids
    Country Sales Manager UK 3 days left
    VERIFIED

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a

    $4158 (Avg Bid)
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    7 bids

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

    $4206 (Avg Bid)
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    7 bids
    LinPhone Development Project 2 days left
    VERIFIED

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    well , we need basically bandwith optimization. in client end they have termination gateways, each gateway can carry 30 simultaneous calls. and all gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should

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    The purpose of this project is to allow a loan officer (client user) at a bank to check the status of his appraisal orders using a custom Alexa Skill instead of logging in to the portal. The Data can be acquired using the web service API Client User Type (UserType 2), GetOrder and GetOrders Methods.

    $980 (Avg Bid)
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    I have an Asterisk server for private use already running, and would like to acid...server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

    $9 / hr (Avg Bid)
    Local
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    we are a coaching company with sports coaching sessions and holiday clubs for children aged 2-11 based around the north west, we have created a website to make it easier for parents that want to book onto our football courses or holiday camps. The search bar that we require for two pages (football sessions and holiday camps) will help parents to find out what venue is closest to them in order for ...

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    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    We'd like to allow a parked caller to press a DTMF which sends the caller to the user that originally parked them. Please provide a full project plan and how much time you'll need to accomplish this.

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $1142 (Avg Bid)
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    23 bids

    ...1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my client where the IP PBX is. I also need to check that the Set nr2 of numbers show the right DIDs when they call external numbers... I need to check that Fail2Ban that was set last

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    1 bids

    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    $53 / hr (Avg Bid)
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    We need a WHMCS addon / Filter For enabling the WYSIWYG editor on product descriptions. Currently you can add certain HTML elements, but we want this WYSIWYG

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $836 (Avg Bid)
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    10 bids

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    49 bids

    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

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    2 bids

    Digital Ocean, Standard Ubuntu VPS server with already configured MySQL server and PHPmyAdmin

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    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    ...business making the purchase. We need an integration that can be used to charge an on-file credit card the 10% fee to the business making the purchase each time a trade is made. Requirements: 1. You will need to determine the best way to integrate credit card processing into the Cyclos platform. 2. You will need to have knowledge of Java applications

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    Local
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    I am looking for a SNOM phone guru that can create a dial plan for me with the following: 4 Digit Internal extension calling (extensions range from 2000-8000 & 9800) 7 Digit Local Calling 10 digit local calling (we are in Canada) 11 digit long distance calling Toll Free 800/855/844/866/877/888 International Calling with 011 Emergency calling with 911

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    A Website already exists, the E commerce needs to be added to the pre-existing HTML website. The site needs to be able to facilitate Online purchases & Deliveries. For more information contact us

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    ...want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL] up to you as you like i just need pass a call on g729 codec

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    add sip trunk to elastix i have sip trunk from STC

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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through

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    i want to add new product to compare iff attributes set matched , so need help with it

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    Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call

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    Hi, I a need simple in usage script that would allow me to to put my mp3 files to one folder, the script should do machine learning and generate music based on the samples from my folder. The script should allow me to choose some basic trainig parameters from command line so that I can compare different settings. The tools should learn from my sample

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    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

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    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    I have posted entries in our Odoo ERP - did not set the "allow reconciliation" to True before posting entries. Need to set all accounts to "allow reconciliation" so that i can reconcile all with the Bank Statements without disturbing existing Entries and Data.

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    Need to develop a ringless voicemail drop system. The system must connect from carrier to carrier to allow VM to be placed on a number without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples

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    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

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    I need an existing website rebuilt. I need you to design and build my personal website.

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    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

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    I’m throwing a sip and paint party. I’m looking for a artist to guide the class, to do a painting

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    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with

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    I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else

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    I require a voicemail d...csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion

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    Hi, I need SMS gateway, Without using any third party services like twilio etc. I need my own gateway. Freelancer must have experience with Voip/PBX/SIP etc

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    We are looking person with specific experties SIP protcol PHP DontNet Also goo din graphics

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    Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)

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    I am looking to create a SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own)

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    ...2nd IP on my voip device i need a solutions to pass call server to local VOIP device i cant use any PC or other device I have only a openwrt Router like TPlink Mr-3420 v2/v3/v5/ i want to run 32calls you can connect server to local a VPN and MASQUERADE connections or you can install any SIP PBX on local route like sofia-sip free-switch and register server

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