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    8,462 sip call software mobile phone 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

    €2700 (Avg Bid)
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    Ich suche jemanden der mir eine funktionierende Verbindung zwischen einem Freepbx und einem Digium 1-Port T1/E1/SIP Gateway (G100) herstellen kann

    €210 (Avg Bid)
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    6 Angebote

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

    €53 (Avg Bid)
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    ...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

    €6935 (Avg Bid)
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    ...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)
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    ...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

    €602 (Avg Bid)
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    ...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 eingebe und die dann den Anruf macht. Später kann man die Funkt...

    €19 / hr (Avg Bid)
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    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.

    €928 (Avg Bid)
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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

    €229 (Avg Bid)
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    SIP register with Custom software 5 Tage left
    VERIFIZIERT

    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [Zur Anzeige der URL Anmelden] i want to run this program on a router Router Link [Zur Anzeige der URL Anmelden] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

    €509 (Avg Bid)
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    5 Angebote
    Programming Bandwith.com 5 Tage left
    VERIFIZIERT

    Help programming sip endpoints, applications voice & sms in [Zur Anzeige der URL Anmelden] I need to connect my [Zur Anzeige der URL Anmelden] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [Zur Anzeige der URL Anmelden] Prior experience

    €8 / hr (Avg Bid)
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    Multichannel Phone Dialer 5 Tage left
    VERIFIZIERT

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

    €588 (Avg Bid)
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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

    €134 (Avg Bid)
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    7 Angebote
    sip to Native dialer APP 4 Tage left
    VERIFIZIERT

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

    €448 (Avg Bid)
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    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

    €736 (Avg Bid)
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    VOIP, SIP Expert 4 Tage left
    VERIFIZIERT

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

    €172 (Avg Bid)
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    Connect IP PBX to Twilio WebRTC 3 Tage left
    VERIFIZIERT

    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

    €37 (Avg Bid)
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    SIP Proxy Server install 3 Tage left
    VERIFIZIERT

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

    €170 (Avg Bid)
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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

    €220 (Avg Bid)
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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Secure Issabel PBX (asterisk) 1 Tag left
    VERIFIZIERT

    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    Trophy icon Création logo Startup 1 Tag left

    ...companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Ca...

    €49 (Avg Bid)
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    95 Einträge

    ...Applications or get one developed at effective cost Click to call features: 1) When number clicked on webpage, viewer types in his/her phone number and submits to call other party, Click to call calls and connects both parties 2) Ability to add an audio file message (in wav or mp3) to the placed call, greeting both parties upon pick up 3) To store voice

    €294 (Avg Bid)
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    ...customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable / turn on a led to indicate a

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

    €159 (Avg Bid)
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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    VOIP ASSISTANCE Beendet left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    Trophy icon JT&T Brand 3 Tage left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

    €193 (Avg Bid)
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    541 Einträge

    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [Zur Anzeige der URL Anmelden] secret GZD397XU Host [Zur Anzeige der URL Anmelden] port 5060 Register String 390415541985@[Zur Anzeige der URL Anmelden]:GZD397XU:390415541985@[Zur Anzeige der URL Anmelden] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain ...

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    ...us. Do you have IAX SIP VOIP Softphone system and apps? We would like to do similar system like www . zoiper . com Zoiper is a phone solution perfectly fit for end users, service providers, call centers or any business willing to benefit from VoIP communications. Zoiper softphone is a standalone client-side software VOIP phone application and is not

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    ...routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new routines quickly and efficiently. We have

    €2120 (Avg Bid)
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    Lightweight SIP Client with Dynamic CRM which would display the customer information. CRM Field and information would display dynamically and action should collect and send to server Database. Required Features: SIP Call with transfer/conference/hold/Mute facility SIP messenger. Dynamic CRM-Dynamic field header and value display for each call according

    €267 (Avg Bid)
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    I need some one who is expert in cyber security & hacking and specially VOIP, Sip, you should be expert in this field.

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    I need professional help with setting up duplicate Avaya voip phone system. I need worker with real experience. I will describe current system i have. I need to buy same equipment, and i need to know what licenses i really need to get or don't need. More details will be given as we talk together. Current System: • Avaya IP Office 500 V2. Build Version

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    as the title indicates, need sip installed on cisco 7960 to work with astersik. total 10 phones

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    Hi. We need to configure asterisk on our ubuntu machine. We need someone high skilled to do the job. We will also need to make it visible over internet with public IP. Also we need to configure some IP phones on it. After all, we need a manual with step by step about how to configure it.

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    I have lan calls working fine, but require someone with Cisco/Asterisk SIP knowledge to help resolve this issue with NAT, no audio not working.

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    I require a voicemail drop software platform. The application will feature voicemail detection function where an outbound call is placed from 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 callcentric

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    Predictive Dialer Beendet left

    ...recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list. There should be a database where agents can comment on the call, this would also need a report with auto report

    €402 (Avg Bid)
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    ...receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default, so just need to initiate a call on gsm

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    ...looking for some one who can make an android mobile app that could make a mobile call trunking application for both directions GSM to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work ...

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    ...verification. The existing code seems to support only provisioning by email address login. Users should instead enter their mobile phone number to register. After receiving an SMS with a verification code the softphone should download the SIP provisioning data from the server. - Create IOS version and deploy to appstore - Create Android version and deploy

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    ...1main server other all asterisk server work like VM host1 IP [Zur Anzeige der URL Anmelden] SIP 5060 DB NAME datastore1 host2 IP [Zur Anzeige der URL Anmelden] SIP 5060 DB NAME datastore2 host3 IP [Zur Anzeige der URL Anmelden] SIP 5060 DB NAME datastore3 host4 IP [Zur Anzeige der URL Anmelden] SIP 5070 DB NAME datastore4 host5 IP [Zur Anzeige der URL Anmelden] S...

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    ...Vonage and Callcentric SIP with our Bitrix24 installation for calling to USA. This will be used with a 10 PC user system for a call center. Other requirements with same: - SIP set up in Bitrix24 - Setting user right for 10 users with extendable option to more users. - Setting up rules for call forwarding, call picking, call sharing etc. - Complete

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