Freelance developer mobile sip dialer jobs

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    12,406 freelance developer mobile sip dialer jobs found, pricing in USD

    ...looking for both individuals or outsourcing companies who could fulfill the requirements below. Requirements 1- Very Good command in English 2- We will provide a reliable dialer (for individuals-Only) 3- Previous sales or telemarketing experience 4- Data are provided with recently updated leads. An Online training would be available + live coaching

    $5 / hr (Avg Bid)
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    8 bids

    Looking for 50 people to generate leads for personal loans for Australians Do you have your own unlimited phone or VOIP with Dialer Do you have your own phone data/Leads are you a Marketer with the ability to create Qualified Leads You will be paid $80 per Settled Personal Loan, Once the Client takes out and is approved for a personal loan you get paid

    $92 (Avg Bid)
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    4 bids

    I need a frontend with asterisk api to open the client's registry when the connection is answered, this connection is generated in principle from a predictive dialer inside the asterisk but can be dialed by the operator also in order to return a contact, and that system you must have the option to import a CSV file of mailing that we receive to dial

    $792 (Avg Bid)
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    20 bids

    Looking for one time setup of auto dialer set up on server.

    $64 (Avg Bid)
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    8 bids

    ...start immediately with agents experienced in Transferring Calls to the Medicare Supplement Call Centers in the United States over a VICI Dialer for over 1 Year at least. Client will provide calling Database,Dialer,and Training. Applying Call Centers Must have minimum 10 agents with experience in Medicare Supplement Call Transfer Campaigns ready to start

    $1150 (Avg Bid)
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    14 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

    $2502 (Avg Bid)
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    20 bids

    ...his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server

    $723 (Avg Bid)
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    3 bids

    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.

    $96 (Avg Bid)
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    9 bids

    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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    16 bids

    ...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 completed

    $36 (Avg Bid)
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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

    ...nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 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

    $45 (Avg Bid)
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    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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    21 bids

    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
    VOIP yate call pass Ending left
    VERIFIED

    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

    $200 (Avg Bid)
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    1 bids

    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

    $602 (Avg Bid)
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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

    $122 (Avg Bid)
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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.

    $75 (Avg Bid)
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    2 bids

    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
    $160 (Avg Bid)
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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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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

    $34 / hr (Avg Bid)
    Local
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    3 bids

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    i have voip device and i have small router it have 32mb ram os Openwrt i 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]

    $216 (Avg Bid)
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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 SIP...

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    Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented...TAPI - Dialling from Windows applications • Text-to-Speech • VoIP Gateways • Voicemail • Visual Indicator for Message Waiting • Voicemail to email • Web Voicemail Interface Mobile data collection to push data into CRM Create reports on the fly.

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

    $1063 (Avg Bid)
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    17 bids

    Following are the requirements: Good Communications Good Internet Speed Goal Oriented Mindset Team Following would be provided : CRM DATA Dialer VOIP Traning Support Team has to adhere to timings for reporting , Meetings, login time .

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    ...screen by App after 2 second. 3. sim selector for calls: mobile has 2 sim, app will read operator names (eg. gp, robi, blink, tt whatever) and store names in app database like sim 1 have (GP or any) and sim 2 have (robi or any). sim will switch calls with integreted built-in dialer based on minutes. its like sim selector before calls based on minutes

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    ...client commission will be your. Centers just have to book appointments. Data will be provided by client. Training will be provided by client Directly in your call center. Dialer and voip provided. Long term campaign. Weekly Payout. Interested Call centers and call center owners contact for more details. - [Removed by Freelancer.com Admin]. Bangalore

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    I want to make an auto-dialer using selenium grid that can call business phones and play advertising commercial

    $74 (Avg Bid)
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    Following are the requirements: Good Communications Good Internet Speed Goal Oriented Mindset Team Following would be provided : CRM DATA Dialer VOIP Traning Support Minimum Five agents are must to login . Team has to adhere to timings for reporting , Meetings, login time .

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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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    ...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 [login to view URL] [login to view URL] [login to view URL] https://www

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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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    18 bids

    I want to make an auto-dialer using selenium grid that can call business phones and play advertising commercial

    $157 (Avg Bid)
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    15 bids

    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.

    $200 (Avg Bid)
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    1 bids

    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 ...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.)

    $5559 (Avg Bid)
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    ...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) ex; not using Google Voice

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    ...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 to local if you want you can install a SIP Signal service on openwrt and pass call to voip device also you can bridge network and assin a server IP on local Voip

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    I need a dialer for automated phone calls

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