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    8,605 android sip client open source code video jobs found, pricing in USD

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

    $311 (Avg Bid)
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    8 bids

    ...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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    7 bids

    Looking for freelancer to configure FusionPBX with PowerPBX hosting. [login to view URL], [login to view URL], 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

    $455 (Avg Bid)
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    15 bids
    Trophy icon Logo design for "Livingston Depot" 4 days 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)
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    118 entries
    Asterisk Expert 4 days 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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    5 bids

    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. [login to view URL] [login to view URL] The 3cx gateway for skype is running only windows 7. and needed skype old version

    $29 (Avg Bid)
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    3 bids

    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

    $30 (Avg Bid)
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    1 bids

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

    $1495 (Avg Bid)
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    7 bids
    ejabberd configuration 2 days left
    VERIFIED

    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 bids

    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 bids

    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 bids

    ...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...

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    27 bids

    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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    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.. [login to view URL] New features need to add 1. SIP Calculator 2. Update Emi Calculator for variable interest rates

    $158 (Avg Bid)
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    33 bids

    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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    ...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 the 4 digit code to a web URI we provide. (c) Prompt user to speak

    $318 (Avg Bid)
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    convert cisco ip phone to sip for working in yeastar s100

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    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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    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 ...

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    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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    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 co...

    $373 (Avg Bid)
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    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
    Sealed
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    13 bids

    Early decay in babies teeth is a big problem and is called early childhood caries...Brush the child's teeth morning and night 4) Bacteria from your mouth can be transmitted to your baby's mouth so don't share spoons etc 5) After a feed clear the mouth with a sip of water 6) don't feed after that 3) don't let the baby breast feed all night or on demand

    $14 (Avg Bid)
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    1 entries

    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 bids

    Add the ability to enter sip trunk credentials to asterisk

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    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 bids

    ...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 bids

    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)
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    ...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 bids

    ...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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    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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    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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    ...check if you are open to work at regular hours for us (much more hours than I propose now). We exist for over 10 years in NL and we have an asterisk based 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

    $20 / hr (Avg Bid)
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    i have installed 3cx pbx and every thing is good then i install sip trunk and configure the did/dod put when call my office number sometimes it ring and sometimes the number are busy need help

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    Hi Dear, We have dial plan for inblond calls . Mean ...NoOp(${ipaddress}) exten=>s,n,Set(ODBC_ADDLIVE(${DID},${CALLERID(num)},${CHANNEL},${UNIQUEID})=1) exten=>s,n,Set(ODBC_CALL_DID_MAPPING(${DID},${UNIQUEID})=1) exten=>s,n,Dial(SIP/${DID}@${ipaddress}, 60, o) exten=>s,n,Hangup() exten=>h,1,Set(ODBC_DELETELIVE(${CHANNEL})=1) exten=>h,2,Hangup(42)

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    you have to install asterisk 1.8 on Puppy Linux make sure asterisk work with sip & iax both also wireguard & TeamViewer you can download OS from here [login to view URL] also if you want you can use Ubuntu or Centos you need to make iso and iso size must be under 500MB i can pay up to 100USD do not ask for more

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    ...will reside on Tizen the function of the app is a bridge between sip client and Phone Native dialer The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Tizen. Must run on bac...

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    We are new small VOIP Services provider and We need a SIP Voip Softphone App for mobiles mainly, eventually for windows too. The App would be similar to Zoiper, Bria, SIPGo, SessionTalk, etc. but customized with our Logo, our proxy domain and most important the capability to use all video and audio codecs including the now free licensed G729.

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    ...web interfaces. To this environment, we now want to find a resource to extend our development team with the following qualifications: Requirement: Good knowledge, be able to code and have references in techniques based on • PHP - Our web applications are built with this language, no actual framework is used today. • JavaScript - with the following technics

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    Seeking somebody that can...accept rooms offered in XMPP XMPP prosody server to be able to handle room authentication + concurrent usage per room previously setup Jigasi dialin number (At the moment SIp registers and enters room but no voice never) Online translation and online transciption Customize mobile app In other words, set it up correctly.

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    ...with reporting our quarterly GSA sales and remittance of the Industrial Funding Fee (IFF) -Acceptance and incorporation of Mass Mod -GSA [login to view URL] catalog uploads via SIP -Assistance with Most Favored Customer discounts to ensure compliance with GSA’s Price Reductions Clause -Assist in preparation for Contractor Assistance meetings performed

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    Required for own numbers, SIP Trunk Server Setup.

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    We need to develop a sip dialer for Windows and Mac.

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    Implement security and failover SIP

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    ...and [login to view URL] need a basic SIP dialer for Android, keeping in mind these features: * PREFERED someone that already worked with Dellmont dialers or similar. * Dialer itself with all the basic functionality of a softphone. * Call history (calls made by the app itself) * Address book * Login/signup activities * Source code aviability There is a pre-compiled

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    We are installing a Cisco 3800 Series Integrated Services Router for a E1 PRI We need someone to configure for the...a E1 PRI We need someone to configure for the following 1. Set Cisco to DHCP 2. Setup Dial peer and Plans 3. Configure the PRI Setting to register on Carrier 4. Register to SIP server , for Inbound and Out bound calling to from the PRI

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    Configure our VOIP setup. We are using a TG100 Yeastar GSM Gateway and Groundwire on iOS. This task is to set them up so they work reliably. Calls from the iOS app will terminate using the SIM. Calls from the SIM will terminate to the iOS app using a push notification. The TG100 Yeastar GSM Gateway is behind a router with a web based interface and port forwarding has been setup.

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    Hello ; We ha...online with there SIP phone on there mobile . For this setup my calls come from another network and we need divert that calls on ext. We can use two methods . 1st : ring all ext. or we can divert calls according our ext. list but no call busy tone and if all ext. busy then call must reject with proper 503 ( rerouted cause code)

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    Configure Avaya Ip Office and Application Server for 10 Users. Program 10 IP Phones. C...Ip Office and Application Server for 10 Users. Program 10 IP Phones. Currently on r8(95) but have upgrade licenses to R11. Equipment is Installed and Wired awaiting set up. Sip Trunk In Place Ipoffice 500v2 Extreme Summit 250 Switch Dell R330 Server J179 Handsets

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