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    8,372 sip web phone windows mobile Gefundene Jobs, Preise in EUR

    ...Schrift Seien Sie bitte bereit, ein erstes technisches Gespräch auf dem Grundverständnis der folgenden Konzepte zu führen: DHCP, DNS, NTP, TCP/UDP, NAT, firewall, switch, VoIP (SIP, RTP). WAS WIR IHNEN BIETEN Attraktives Gehaltspaket mit Weiterbildung- und Entwicklungsmöglichkeiten Internationales Team Smart Working (2 Tage in der Woche) [Removed by

    €2672 (Avg Bid)
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    1 Angebote

    Ich suche jemanden der mir eine funktionierende Verbindung zwischen einem Freepbx und einem Digium 1-Port T1/E1/SIP Gateway (G100) herstellen kann

    €208 (Avg Bid)
    €208 Gebot i.D.
    6 Angebote

    Configure A2Billing + Elastix 2.5 (Using DAHDI & SIP Trunk Provider)

    €52 (Avg Bid)
    €52 Gebot i.D.
    2 Angebote

    ...herzustellen ( es zählt die Zeit bis die Leitung besetzt ist ). Die Gegenstelle ist in 90 % der Fälle ein GSM Teilnehmer. Wie die Verbindung Aufgebaut wird ( z.B. Asterisk mit SIP Gateway oder GSM Dongles ) wird nicht vorgegeben solange das Ziel erreicht wird. Die Aufgabe ist erfüllt wenn ein Nachweis erbracht ist, dass in 100 Bespielanrufen ( verteilt

    €6861 (Avg Bid)
    €6861 Gebot i.D.
    2 Angebote

    ...Tehnologie entwickeln unsere Experten innovative IT-Lösungen und anspruchsvolle Produkten. Für unsere Kunde suchen wir eine/n SIP/WebRTC Freelancer. Ihr Profil: Erfahrung mit WebRTC. Sehr gute Verständnis für TCP/IP und SIP/RTP Protokollen. Informatikstudium oder informationstechnische Berufsausbildung. Mindestens drei Jahre einschlägige Beruf...

    €30 / hr (Avg Bid)
    €30 / hr Gebot i.D.
    1 Angebote

    ...entweder über vorprogrammierter Schleifen oder noch besser per Schnittstelle (Befehle unseres System an den Telefonserver), eine Reihe von Personen auf Festnetz/Mobil anrufen (kein SIP-Telefon und Ausland). Beim annehmen des Anrufst wird eine kurze Bandansage abgespielt. Bleibt er in der Leitung (quasi als Bestätigung, dass er Zeit hat) wird er im First-Come-First-Serve

    €596 (Avg Bid)
    €596 Gebot i.D.
    1 Angebote

    ...haben einen lokalen PBX (Asterisk mit Elastix) im Netzwerk, der über eine IP [Zur Anzeige der URL Anmelden] erreichbar ist. Auf XOS und Windows benutzen wir Zoiper und x-lite um mit dem PBX zu kommunizieren. Leider gibt es für Chrome OS noch keinen SIP client. Diesen müssen wir entwickeln. Zu beginn reicht eine einfach Erweiterung, in die ich die Telefonnummer e...

    €19 / hr (Avg Bid)
    €19 / hr Gebot i.D.
    1 Angebote

    Ich benötige GoAutodial bzw. Vicidial auf einem Server fertig Installiert haben. Es soll ein SIP-Trunk angeschlossen werden und eine API zu ZohoCRM gebaut werden. Für weitere Fragen stehe ich gern zur Verfügung.

    €918 (Avg Bid)
    €918 Gebot i.D.
    4 Angebote
    Country Sales Manager UK 4 Tage left
    VERIFIZIERT

    ...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

    €3615 (Avg Bid)
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    7 Angebote

    ...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

    €3657 (Avg Bid)
    €3657 Gebot i.D.
    7 Angebote
    LinPhone Development Project 3 Tage left
    VERIFIZIERT

    ...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 --> ready in four weeks

    €2151 (Avg Bid)
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    18 Angebote

    ...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

    €629 (Avg Bid)
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    3 Angebote

    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.

    €83 (Avg Bid)
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    9 Angebote

    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.

    €8 / hr (Avg Bid)
    Lokal
    €8 / hr Gebot i.D.
    14 Angebote
    Freepbx and Asterisk settings 2 Tage left
    VERIFIZIERT

    ...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 comple...

    €31 (Avg Bid)
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    4 Angebote

    ...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 --> ready in four weeks

    €979 (Avg Bid)
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    22 Angebote

    ...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

    €39 (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.

    €45 / hr (Avg Bid)
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    19 Angebote

    ...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 --> ready in four weeks

    €727 (Avg Bid)
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    10 Angebote
    VOIP yate call pass 1 Tag left
    VERIFIZIERT

    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

    €174 (Avg 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

    €523 (Avg Bid)
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    Wordpress Plugin for Phone Call 18 Stunden left
    VERIFIZIERT

    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

    €107 (Avg Bid)
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    9 Angebote
    Actionscript 3.0 SIP 13 Stunden left
    VERIFIZIERT

    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.

    €65 (Avg Bid)
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    PHP SIP client 11 Stunden left
    VERIFIZIERT

    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
    €139 (Avg Bid)
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    VOIP Project 8 Stunden 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

    €1168 (Avg Bid)
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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

    €30 / hr (Avg Bid)
    Lokal
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    3 Angebote

    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

    €217 (Avg Bid)
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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 [Zur Anzeige der URL Anmelden] or yate or besip [Zur Anzeige der URL Anmelden]

    €188 (Avg Bid)
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    add sip trunk to elastix i have sip trunk from STC

    €16 / hr (Avg Bid)
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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...

    €33 (Avg Bid)
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    Looking...• Roaming Extensions • Route by Caller ID • SMS Messaging • Spell / Say • 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 t...

    €787 (Avg Bid)
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    we wish to have someone connect and configure our [Zur Anzeige der URL Anmelden] to our SIP and Trunk

    €924 (Avg Bid)
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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。

    €45 (Avg Bid)
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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

    €1026 (Avg Bid)
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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

    €497 (Avg Bid)
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    15 Angebote

    ...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 [Zur Anzeige der URL Anmelden] [Zur Anzeige der URL Anmelden]

    €534 (Avg Bid)
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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

    €23 / hr (Avg Bid)
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    18 Angebote

    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.

    €174 (Avg Bid)
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    1 Angebote
    Design project Beendet left

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

    €72 (Avg Bid)
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    20 Angebote
    webrtc asterisk Beendet 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

    €261 (Avg Bid)
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    1 Angebote

    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

    €156 (Avg Bid)
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    3 Angebote

    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

    €56 (Avg Bid)
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    2 Angebote

    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 [Zur Anzeige der URL Anmelden] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

    €427 (Avg Bid)
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    5 Angebote

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

    €402 (Avg Bid)
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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

    €149 (Avg Bid)
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    We are looking person with specific experties SIP protcol PHP DontNet Also goo din graphics

    €362 (Avg Bid)
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    6 Angebote

    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.)

    €4833 (Avg Bid)
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    ...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 or anything like that. It must be able to send and receive SMS messages from a standard cell phone. The overall price is negotiable

    €511 (Avg Bid)
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    19 Angebote
    €12 Gebot i.D.
    11 Angebote

    ...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

    €489 (Avg Bid)
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    6 Angebote