Sip simple asterisk presence jobs

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    17,210 sip simple asterisk presence 找到的工作,价格在 USD

    Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.

    $1532 (Avg Bid)
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    3 竞标

    VoIP Video IM等即时通讯功能。 支持Android,IPhone,Window Phone等客户端应用。 熟识SIP Webrtc XMPP。 提供Demo。

    $5555 (Avg Bid)
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    1 竞标

    主要指导elastix asterisk 的安装配置工作,熟悉Asterisk的模块(包括体系结构,配置文件,Log日志,History),能重新打包ISO文件

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    5 竞标

    ...8、需要提供所有源代码和演示版本(演示版本的运行平台包括android、IOS),以及可交付的中转服务器; 9、并不需要完整的语音通讯软件,只需要能够实现语音传送、播放的功能模块即可。 10、演示内容:A用户点击“通话”按钮,中转服务器可通过设置好的IP地址,找到B用户,并发送通话请求。B用户接受请求后,双方可以进行通话。 11、你可以采用第三方组件,例如SIP,但是需要告诉我们是哪家的,效果如何。因为考虑到我国网络问题,所以建议是自己搭建服务器。 12、我们已经开发的软件可以在安卓、苹果商店找到,叫IEMAKER,通过安卓盒子,可以实现直播,但是目前还不能实现声音直播,因此,需要开发这个模块,用于该软件。

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    i have a freepbx system and need to configure one sip detail but not able to use that details like as sip trunk. if i put same details on xlite or IP phone its working fine. so its a simple task but i need some expert one for this.

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    i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done

    $53 (Avg Bid)
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    Softphone WEBrtc - asterisk 6 days left
    已验证

    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

    $316 (Avg Bid)
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    ...CarrierBid. I have 40 more icons that I need designed for the following subjects: 1. Free WiFi 2. WiFi Router 3. Network Access Point 4. Customers walking 5. Just the word: “SIP” with a telephone handset 6. Paper icon with the words: RFP RFQ RFI with a $ symbol 7. Language Translation services 8. iphone notification 9. Increased sales 10. Advertising

    $250 (Avg Bid)
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    ...available, issue a tone. when call has been answered issue a tone, when call gets hung up, issue a tone. (each must be different tone) On phone 1 Create a sip client Client will register to a sip server and receive calls with destination phone number. When the client receives a destination phone number, it will send this to its headset speaker as a

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    I need some help with internet marketing.

    $21 (Avg Bid)
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    Asterisk configuration on aws 5 days left
    已验证

    I need to configure asterisk on amazone aws and make call from outside. Please bid only if you ready to work right away over team viewer. Thanks

    $39 (Avg Bid)
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    Instalar una centralita basada en asterisk en nuestro centro con 10 extensiones y IVR.

    $18 / hr (Avg Bid)
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    6 竞标

    ...want to enhance the meme, including a background image (see attached samples) Choose from one of the options below to create your meme for this contest (do not include asterisk in design): *FACT: Social media platforms make money from YOUR usage through ad sales. FACT: You can legally SHARE in this revenue! Comment "more information" to learn

    $25 (Avg Bid)
    担保
    $25
    28 作品

    Necesitamos asesoria de como instalar un prouyecto asterisk para usarlo con webrtc y sipjs

    $265 (Avg Bid)
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    6 竞标

    Zoho's phonebridge plugin supports up to Asterisk 1.4 [login to view URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially

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    FreePBX Admin 2 days left

    Hey, guys, I have a simple project. I have a FreePBX server that I need to build an inbound Sip trunk to 2 separate carriers + build inbound routes for them. It should be a simple process. Please respond to the bid with "What up Dingo" at the beginning of your message so that i know you have read.

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    5 竞标
    VOIP Expert needs -- 2 2 days left
    已验证

    I have installed my amazon ubuntu server and asterisk server . Asterisk server is working well . I am looking for somone that can install and connect SIP so that I can send sms code perfectly

    $128 (Avg Bid)
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    4 竞标

    I have installed my amazon ubuntu server and asterisk server . Asterisk server is working well . I am looking for somone that can install and connect SIP so that I can send sms code perfectly

    $472 (Avg Bid)
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    1 竞标
    pfsense expert -- 3 1 day left
    已验证

    ...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud

    $94 (Avg Bid)
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    3 竞标

    Need sip calculator script written in php

    $10 / hr (Avg Bid)
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    12 竞标
    C++ Programing 1 day left

    ...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck

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    Web development 19 hours left

    I need a new website. I need you to design and build my online store. Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, busines...Service provider company:- - Credit report, - Home loan, mortgage loan, personal loan, business loan, gold loan, car loan - investments, fixed deposits, insurance, mutual fund, SIP.

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    The business is a provider of cloud based communications which includes: Skype For Business Microsoft Teams Cloud Contact Centre Office 365 Sip Trunking Hosted PBX We require assistance in developing and revamping our website content, feature guides, product bundles, resources for customers (How2Guides/FAQ etc) some of which we have but need updating

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    I have all details .i need to configure sip and configure asterisk server . VIP experts

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    We are looking for an expert in Globus toolkit 6. Configuring asterisk, postfix, red media server and QGIS in Ubuntu server.

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    Need software for making vx670, ingenico 5100 or vx 510 working in offline mode/ force offline mode (including Operating System) without presence of card. Already have a vx670 POS. The VX670 should have the following function that meets my needs: [login to view URL] on the device [login to view URL] down to menu item "offline" or "manual offline" or "force" ...

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    A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign

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    Hello i need to bulid sip filter to filter Retail Voip traffic

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    7 竞标

    We are looking to develop a mobile a...integrate specific API from our middle ware platform. Initially the app will only need to be for android devices. The mobile will need to integrate with a telephony platform for sip communication and a messenger platform. The majority of the mobile devices will not have cellular service and will be strictly wifi.

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    outbound calls and select options automati, will send more info. about project .

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    I would like to have our Church to get a website created that will cater for the following:- - Have an online shop - Accept sponsors to activities - Have a contact form - Have a login for member registration - Social media links - developed in 20 days.

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    hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system

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    Looking for assistance in configuring Cisco call manager for sip trunking.

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    hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly

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    Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones

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    We want a logo/visual presence that express these factors: - internal knowledge development - network of professionals - effective learning - collaboration - dynamic loose coupling between peers - target sectors for the network: education, childcare, health - target participants in the network: leaders, professionals (eg. teachers, nurses) The logo/visual

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    ...person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not

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    I need a very simple customisation of a Android based free / open source SIP client(sCIPsimple,Linphone or anything). I want the default values of my server to be populated in place of blanks in configuration forms. So, all you need to do is replace the values and compile and give me.

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    I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]

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    Hello. We are looking VOIP Tunnel software solutions to bypass voip calls in blockage area . It is encrypts/decrypt SIP and RTP packets and send voip traffic between soft switches without blockage . We are looking experienced guy who already developed or worked on this .

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    We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.

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    21 竞标

    Hello I need someone to help me install whitelist on my asterisk white will reject all the calls with 503 and pass only the numbers in the list

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    14 竞标

    Need to configure 10 internal accounts on IP phones, connect 3 external telephone lines. Set the voice greeting and the voice menu according to the technical task. Implement a CDR report and record conversations.

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    ...initially provide reporting capabilities on Twilio's SIP trunking product. Using Twilio's REST API, you'll build the following: 1. User auth 2. Dashboard with usage data (much like what twilio provides now, but for the subaccount) 3. Historical reporting - whatever twilio provides data-wise on the SIP trunk side of the house (call date/time, ANI, duration

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    I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a

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    I need an application that runs on Android phones (4.0 or above) and can send/receive calls 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. Full description below. Please review and message me with considerations. If any

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    Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !

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    Can you make android and iOS app for Asterisk client? Similar as Zoiper for iPhone.

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