Mobile sip voip dialer project jobs

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    16,571 mobile sip voip dialer project jobs found, pricing in USD

    ...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

    $1162 (Avg Bid)
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    6 bids
    Gsm voip gatway 6 days left
    VERIFIED

    I need an Android app. I would like it designed And Build Android application make mobile phone act as GSM to voip gateway To connect with Asterisk PBX We needs to use an android cellphone as an asterisk channel/Gateway. His aim is to make phone calls using his Asterisk PBX through a cellphone running android. It aim to make phone calls using

    $1132 (Avg Bid)
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    14 bids
    Country Sales Manager UK 5 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

    ...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 you have sales projects in your project history or that you can refer to experience outside of Freelancer

    $4206 (Avg Bid)
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    7 bids
    VoIP telephony 5 days left
    VERIFIED

    We have iPECS-100 behind Chekpoint firewall. All customers use soft phones clients. Have periodically experience communication problems. Need a person for a one-time or permanent job

    $51 (Avg Bid)
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    4 bids
    LinPhone Development Project 5 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

    $2919 (Avg Bid)
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    18 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

    $988 (Avg Bid)
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    2 bids

    I have an Asterisk 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
    android voip call application 4 days left
    VERIFIED

    System requirements Registration screen Logon screen online Users screen Server php and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure and encrypted between the parties Simple appli...

    $93 (Avg Bid)
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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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    11 bids
    Freepbx and Asterisk settings 4 days left
    VERIFIED

    ...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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    4 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

    $1126 (Avg Bid)
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    22 bids

    ...and would like to offer you my project. We can discuss any details over chat. Below a summary.. I have a IP PBX with 2 sets of 10 numbers. Set 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

    $45 (Avg Bid)
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    1 bids

    i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

    $166 (Avg Bid)
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    8 bids

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

    $52 / hr (Avg Bid)
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    18 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 2 days 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
    app development 2 days left

    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
    VOIP phone system 2 days left
    VERIFIED

    we need a phone system based on twilio platform.

    $652 (Avg Bid)
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    21 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

    $123 (Avg Bid)
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    Actionscript 3.0 SIP 2 days left
    VERIFIED

    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.

    $30 - $250
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    0 bids
    PHP SIP client 2 days left
    VERIFIED

    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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    VOIP Project 1 day left

    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

    $1343 (Avg Bid)
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    14 bids

    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
    Create SNOM Phone Dial Plan 1 day left
    VERIFIED

    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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    4 bids

    Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

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    20 bids

    ...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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    ...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 Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer &bu...

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    I want to use raspi as gsm gateway. This is the basic task. Details will be explained.

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    8 bids

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

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    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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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 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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    ...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,india

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

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    Voip Calling App with unlimited Bandwidth, It will Be used for calling East African Countries, it must be functional.

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

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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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    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 have installed Dinstar GSM gateway, but I have voice quality problem. Who can solve it, please contact with me

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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 drop ...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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