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    8,372 use android phone sip gateway pekerjaan ditemukan, seharga USD
    Softphone android Berakhir left

    Un softphone que maneje cuentas SIP para ANDROID

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    Saya Pernah Direkrut Ngentry Dikantor Badan Stastistik Itu Terjadi Tahun 2015 . Jam Ngentrynya Dimulai Jam 12 Siang Sampai Jam 4 Pagi , Kebetulan Saya Sip Jam 9 Malam Sampai Jam 4 Pagi .

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    setting IP PBX Berakhir left

    jasa setting untuk keperluan kami yaitu: Billing Syste...System / Budgeting per ID / User Recording All Call IVR / Auto Attendance Call Fax to Mail & Mail to Fax Bisa Cabang IP PBX Yealink BLF [Busy Lamp Field] Sip Messenger connect Android APPS VOIP saat ini kami memakai program elastix dengan hardware openvox a814hj0582xk dengan chip 8fxo

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    ...Care untuk bekerjasama dengan PT Kronus Indonesia - Perawat, Fisioterapi, Okupasi Terapi, Terapi Wicara & Bidan - Laki-laki dan Perempuan - Usia 21th – 45th - Memiliki STR/SIP yang masih berlaku - Mampu bekerjasama dengan Team / Independent - Penempatan wilayah Jakarta dan Luar Jakarta (Bandung, Semarang, Yogyakarta, Surabaya, Medan, Makassar atau

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    this program for information tcooperation in Goverment "Pemerentah kabupaten WONOGIRI"(Dinas Koperasi dan UMKM) INDONESIA, Can use for ALL Dinas koperasi and UMKM ind Indonesia

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    Country Sales Manager UK 4 Hari left
    TERVERIFIKASI

    ...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 Country

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    ...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

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    LinPhone Development Project 4 Hari left
    TERVERIFIKASI

    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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    ...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 send to specific termination . call should pass with sip , g729 or g723. in

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    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.

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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.

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    Lokal
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    Freepbx and Asterisk settings 3 Hari left
    TERVERIFIKASI

    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 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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    ...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

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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.

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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

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    VOIP yate call pass 1 hari left
    TERVERIFIKASI

    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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    app development 1 hari 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

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    Wordpress Plugin for Phone Call 1 hari left
    TERVERIFIKASI

    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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    Actionscript 3.0 SIP 20 jam left
    TERVERIFIKASI

    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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    PHP SIP client 18 jam left
    TERVERIFIKASI

    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 14 jam 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

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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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    Lokal
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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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    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 untuk melihat URL] or yate or besip [login untuk melihat URL]

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

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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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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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    Ringless Voicemail Berakhir left

    ...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 [login untuk melihat URL] [login untuk melihat URL]

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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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    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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    Design project Berakhir left

    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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    webrtc asterisk Berakhir left

    Necesitamos hacer un softphone basado en WEBrtc, que funciones desde todos los navegadores compatibles con esta tecnología, se conectan por sip a nuestro servidor asterisk

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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...a 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 untuk melihat 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 untuk melihat URL] to a software that was develop for window [login untuk melihat 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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    I need SMS Gateway Berakhir left

    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.)

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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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    ...connections and USE 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

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    i want to setup an gsm voip gateway GUI using asterisk+a2b+chan dongle, use huawei e173u-1 and e153u-1. i already have a running server, but i want to setup a new server with the following features in a GUI: 1-sim card suspension detection based consecutive unsuccessful number of calls and discard(dongle stop now) the suspended ones as well as not

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    ...the following: 1- Write a script for VICIDIAL installation with a single command line from [login untuk melihat URL] 2- Tutor the proper basic setup for interconnecting with SIP trunk 3- Tutor on how to use VICIDIAL in the following concepts: a- Press-1 campaign: where we will send a pre-recorded message that has an offer, if the customer is interested, he presses

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    XMPP-based client application integrated with a SIP-based client application.

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