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    15,956 sip client iphone 2009 仕事が見つかりました。次の価格: USD
    Build an app 6 日 left

    ONLY APP DEVELOPERS FROM INDIA REQUIRED FOR THIS PROJECT. make an interactive app for IOS and android. app is airshow event based and has to be i...Date / time of entry & exit from the event venue c. Mode of receiving any communication / feedback generated by the user. 5. Timelines: app should be ready for launch by July 2009 after Beta testing etc.

    $2353 (Avg Bid)
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    82 入札

    vicidial expert, to create a sip trunk with an external sip provider

    $111 (Avg Bid)
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    6 入札

    We are looking for someone who has good experience in video chatting app and H.323, SIP. It is very urgent project, so if you don't have good experience in this field, please don't waste our time. Thank you.

    $23 / hr (Avg Bid)
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    16 入札

    sip VoIP dialer with rcs futures voip call sms chat video ex...

    $18 / hr (Avg Bid)
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    4 入札

    I need a VB script that will extract specific information from the website into a...instructions can be found in the attachment. I need the extraction to be as quick as possible. All seasons and all pages will need to be picked up. Only pick up data from 2008/2009 to 20118/2019. Note the code will need to be adaptable for when new seasons start

    $14 (Avg Bid)
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    5 入札

    i need someone that can login to my computer and setup my fusion PBX should accept calls from vitelity SIP Provider, and also to make calls thru my outgoing provider.

    $173 (Avg Bid)
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    9 入札
    Robocall server 4 日 left
    認証完了

    A center of family planning is looking for a Robocall solution to remind patient to visit doctors, etc. Informa...reminder them the date of the visit. at the end of each day, a full report should be given. The voices will be prerecorded by our team here. We will use goip gsm gateway (sip) for the calling. We have servers to host the application.

    $263 (Avg Bid)
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    6 入札

    ...positive response from client > contact. 4. Phone book exist around 2 millions name/address/phone number in excel/CSV/mysql or any other format. 5. Dial #1 – simple dial surface, numbers, #, * etc. all via PC/laptop windows 10. NO landline or mobile phone for dialing only the PC and PC microphone + PC headset. Each user has own SIP account and should use

    $1717 (Avg Bid)
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    17 入札

    I need a simple softphone that allow to SIP register on a freepbx server. With just the dialer, not accessing phone contacts and all the features that most softphones comes. It must run in the background when closed so it can receive calls. Guess is just that, hope your offers! Thanks

    $204 (Avg Bid)
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    6 入札
    Goautodial specific functions 2 日 left
    認証完了

    ...function: when agent click on button, the client can enter credit card details by DTMF and get back to agent to continue discussion 2) Regarding any number in any list: show a different calling number on customer phone for each attempt. Will give you 10 calling numbers that we have 3) Set up load balancing on 3 trunks (sip provider). When dialing on lists,

    $283 (Avg Bid)
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    6 入札
    Digital logo 2 日 left

    I'd like to create a logo for my website and work as an author and speaker on personal development. My initial idea is a play on my initials (SIP) and I've attached a sketch of versions that I have used. Would like someone to play around with that and come up with something professional/digital.

    $62 (Avg Bid)
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    56 入札

    We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.

    $578 (Avg Bid)
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    3 入札

    Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX

    $143 (Avg Bid)
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    2 入札

    ...kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC.mydomain.com-onnet_1-b2b_1' with owner SIP/telsam_ip-00003486 in place (Method: BYE). Rescheduling destruction for 10000 ms. if you interested in I can share you more information. and budget is negotiable fixed or hourly

    $100 (Avg Bid)
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    1 入札

    We need an expert in OpenSIPS that can change the SIP response of 200 ok to something else or delay it full details will be given later.

    $325 (Avg Bid)
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    4 入札

    ...allow only Checked In guests to sign in using surname and Room number. once signed in also allow guest to connect to the Hotel WIFI also within the APP we will need a build SIP/VOIP on the App which will interface with hotel PABX and the app will become the guest extension as long as his signed in and for the duration of his stay. This app will need

    $2499 (Avg Bid)
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    137 入札
    WebRTC / SIP expert 8 時間 left
    認証完了

    The client is using astrix server. They will give us a vpn connection to their network And we need to setup WebRTC client to talk to astrix server. We need to use open source functionality working which it can't currently get a function to work within a type. [ログインしてURLを表示]

    $20 / hr (Avg Bid)
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    28 入札

    Want to start a VOIP/calling card solution ,need to install the necessary software (asterix,freepbx,a2billing etc).Advice the best solution

    $264 (Avg Bid)
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    16 入札

    ...required, and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network

    $13 / hr (Avg Bid)
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    8 入札

    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [ログインしてURLを表示], [ログインしてURLを表示], Debian ...Plans, PBX Extensions, I/O Trunks, Voicemail, DISA, IVR & local number display on the go, Call Forwarding, Call waiting, Linphone setup and instructions, call recording, setup sip trunks. for 4 - 6 lines. PHP or LUA scripting, VOIP, interconnect

    $462 (Avg Bid)
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    16 入札
    Trophy icon Logo design for "Livingston Depot" 終了 left

    "Livingston Depot" is the Name we need a logo for.. - "Sip & Shop Station" is the Tag Line This is a boutique cafe that is located in Livingston, Tennessee. This shop offers local artists, hand crafted items for sale, with a small cafe offering beverages, and lunch items. The keyword inspirations are elegance, down home, boutique, station, depot

    $40 (Avg Bid)
    保証
    Asterisk Expert 終了 left

    ...and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system

    $17 / hr (Avg Bid)
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    6 入札

    I need a call with multi skype user via SIP. and i saw 3cx gateway for skype program. this product is no longer available for download nor maintained. [ログインしてURLを表示] [ログインしてURLを表示] The 3cx gateway for skype is running only windows 7. and needed skype old version

    $29 (Avg Bid)
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    3 入札

    must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account

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    1 入札

    We need an implementation of the services mentioned in the subject. Thanks.

    $1495 (Avg Bid)
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    7 入札

    I need help you configure my ejabberd server with sylkserver for translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

    $22 / hr (Avg Bid)
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    2 入札

    We have a multi-tenant PBX system and need a simple SIP phone provisioning tool written in Javascript Node.JS, using Angular or React, running on a MEAN/MERN stack. This will not be integrated with the PBX, however our B/OSS system will add, suspend, and delete tenants and extensions to this new system you will create via a REST API. A commercial

    $198 (Avg Bid)
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    5 入札

    Setup a VoIP server for termination the requirements are: 1. SIP connections to gsm gateways through VPN is required 2. Billing software 3. Load balance traffic to gateways 4. setup connections with 3 clients

    $238 (Avg Bid)
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    7 入札

    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

    $178 (Avg Bid)
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    27 入札

    I am looking for an experienced FreeSwitch engineer who can work with latest freeswitch version with SIP Trunks to do following tasks I will be passing destination number , CLI , ring duration , max call duration , audio file , unique call id to Freeswitch (JSON API) , the dial plan should start dialing numbers and plays a voice file when the call

    $1272 (Avg Bid)
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    5 入札

    I have already built app on play store called "Loan Emi Calculator". Please review the app befo...Calculator". Please review the app before you bid, here is a store link.. [ログインしてURLを表示] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

    $158 (Avg Bid)
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    33 入札

    I have an existing Shoretel system with a PRI switch. Using the SETU VTEP device I want to replace my existing PRIs with SIP trunking by using the SETU VTEP as a gateway. I need help programming the device to integrate with my existing Shoretel system.

    $200 (Avg Bid)
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    1 入札

    1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post

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    8 入札

    convert cisco ip phone to sip for working in yeastar s100

    $125 (Avg Bid)
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    8 入札

    Need to Configure 2TATA Sip Trunk line on my freepbx via remote desktop like TeamViewer. FreePBX hardware is assembled and installed freepbx.

    $18 / hr (Avg Bid)
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    2 入札

    Sip client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip ...

    $245 (Avg Bid)
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    5 入札

    Hi, I'm looking for someone to connect a twilio number that we currently have to a sip endpoint thats already configured for voice. We are using the X-lite Voip client and would like to get SMS functionality enabled on it.

    $39 (Avg Bid)
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    5 入札

    SIP CLIENT FOR RASPBERRY PI3 this software to be developed for Linux using Python. SIP client for Raspberry pi3 We need to make sip client for raspberry pi3 on parking system machine, Software must have source code in python and auto start with raspberry linux SIP client must call just one number that we insert in the code, over GPIO inp...

    $373 (Avg Bid)
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    5 入札

    Looking for someone knowledgeable on freepbX and mobile application development to create a VOIP application for customers service . The reference app is named as "smart line" You can check the app's features . I need the exactly same features with some new designs .

    $6 - $11 / hr
    シール
    $6 - $11 / hr
    13 入札

    Set up sylkserver to translate jingle/sip between xmpp client and SIP proxy and/or PBX. Goal is that XMPP client can make a call through the PBX to the outside work (as a regular UA) and that the XMPP client can be called via the PBX.

    $21 / hr (Avg Bid)
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    8 入札

    Add the ability to enter sip trunk credentials to asterisk

    $100 (Avg Bid)
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    1 入札

    Please visit the link [ログインしてURLを表示] where you find many posts (permalinks) from 2009 until today. Automate a process which opens the blog post itself and saves all the images in highres (1'280px in width lightbox mode). The images should be named with the description at the bottom. If there are multiple images (which is always

    $65 (Avg Bid)
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    8 入札

    I am looking for a senior architect/developer to design and develop gsm telecom solutions based on ss7 or diameter or sip communication protocols. The scope will be discussed later, the pay will be very good

    $8985 (Avg Bid)
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    15 入札

    ...writer to write an original story featuring a seductive buxom woman who's big unwieldy breasts contain a milk that will wickedly shrink the size of any man who drinks it. One sip is enough to make him a midget and too much will send him down to bug size or smaller. Find a way to work this weird concept into a sexy short story with a full plot and I'll

    $130 (Avg Bid)
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    23 入札
    Trophy icon Need a logo for a company beverage 終了 left

    Hello, I need a logo for a beverage company. The name is "Soufrière Hills" The tag line is “every sip makes you family and comforts your soul.” Please use the following hex codes only: #549127 #10375d #edc857 Use all three or two colors in any combination. Please be as creative as possible. This logo is for a beverage that is from the island of

    $30 (Avg Bid)

    ...minimize the BW in client side with good quality of voice. We need some kind of bandwidth compression system ( up to 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of the scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using

    $1742 (Avg Bid)
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    6 入札

    ...simple sip to tapi tsp app with simple features like sip registration like an sip client no voice or codec are needed the incoming call must only transmitted to tapi tsp driver with caller in number the out bound calls will be managed with action url so from any application how supports tapi functionality can dial out by using the tapi via sip to his

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    10 入札

    Hello, we are selling fasteners in germany. We give you a list of 1...to 3pm GMT. Also part time is possible. You need to speak very good and clear english. Please bid for 1000 Calls/contacts. I will provide a phone account to use. Skype or a SIP Account, so that you dont have costs. When you are good, this can be be a project for much more contacts.

    $590 (Avg Bid)
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    41 入札

    SIP intercom broadcast software View online status View the status of the call Setting up a voice recording Call management Single-key talks, multi-party meetings Surveillance, monitoring, shouting and deterrence Arbitrary Paging and Interphone Recording Talk queuing, strong

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    5 入札

    ...cloud PBX. We sell this as a service to companies and resellers. It is based using our own heavily customised linux distro. We also want to experiment with jingle (xmpp client) and SIP integration using a combi of sylk server and opensips. IMO a real challenge for experienced engineers. I can be reached by mail at max@axeos.com. We can start with some

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    1 入札