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    10,408 sip phone cell phone jobs found, pricing in USD

    I need an Asterisk setup with full Call Center features including sip trunk for outbound campaign and inbound call. More details will be provided to the qualified freelancer

    $150 (Avg Bid)
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    3 bids

    A mobile dialer app for modern mobile interfaces that should work with any sip voip server using sip protocol and support all the standard codecs, consuming low bandwidth, voip dialing using wifi, 3G/4G and have the functionality to use local minutes (as calling card). user friendly and quality voip calls. User may registered through his verified mobile

    $702 (Avg Bid)
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    35 bids

    ...webpage, and the colour of most text. List will be provided as a spreadsheet (Excel or Google Docs), to be filled in with Hex Colour codes, captured using an application like Sip. We will provide several examples where we have captured the colours already, and will check in and verify work after the first 5-10 to make sure that you are getting the

    $226 (Avg Bid)
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    19 bids

    I need a sip-phone (IOS, Android, Windows) able to register into Asterisk and peer a SIM in a GSM gateway. Sip Client should be able to send/receive voice calls, SMSs and USSDs. Sip Client should be able to top-up the peered SIM.

    $1218 (Avg Bid)
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    10 bids

    We have conversations (see githup) as basic and inside this we need some modifications. We run our own xmpp server and separates sip server with csipsimple. By api the registration in conversation is there one sip and xmpp users. Under the conversations user can call other user in his contact list. For this there is already some working examples.

    $1358 (Avg Bid)
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    18 bids
    Cell ID App 5 days left

    ...search result accuracy. 2G/3G/4G serving and neighbor cells information measurement SPOT & NEIGHBOUR CGI. - Cell ID - LAC - MCC - Signal Strength - Latitude an Longitude of the Cell Tower - Distance between the device and the cell tower - Export recorded data to XML, KML (Google Earth), or CSV files. - App to SD card - Sound and Vibration

    $185 (Avg Bid)
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    2 bids

    i need an artist to teach a class at a paint and sip I'm hosting on Feb 10th for 2 hours

    $107 (Avg Bid)
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    7 bids

    I will give access to a linux box, you can install asterisk along with any other services, I want a web interface, where i will copy paste 200 phone numbers, also there should be option to upload a voice record (audio) , once i click play button. Each of the 200 numbers should get a call from their own numbers (spoof) simultaneously and all of them

    $504 (Avg Bid)
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    3 bids

    Skill Pre-Requisites: Knowledge of WebRTC Gateway and not WebRTC web-client. Have developed SIP Servers applications (SBC or PBX) Good knowledge in WebRTC, Web Sockets, HTTP Knowledge of Asterisk server and Jitsi Media. Experience in product development and system engineering for WebRTC. Knowledge on JAVA will be good to have (Not mandatory)

    $19 / hr (Avg Bid)
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    4 bids

    We have our own sip / xmpp server and we have compiled the latest build of csipsimple and Linphone. We want a secured voice over zrtp. This is a peer to peer encryption but i dont get it work, not under csipsimple and not under Linphone. I need support to get this work. Linphone and csipsimple can be downloaded on playstore.

    $157 (Avg Bid)
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    1 bids

    Need all code and all resources associated with it delivered, hopefully in Visual Studio with mysql backend. Open to other suggestions on codebase. [url removed, login to view]

    $173 (Avg Bid)
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    8 bids

    hi, i need to configure trunk sip for outbound campaign and inbound call. im stuck in dialplan setting. in asterisk debug i can show all circuits are busy. i wait your reply tanks

    $21 / hr (Avg Bid)
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    14 bids

    Call functions like mute, conference, hold, transfer, and call rec...integration from LDAP, Outlook, or CSV Low resource consumption Supports SIP, XMPP, and IAX accounts TLS, SRTP, and ZRTP encryption Voice Codecs: G.729 (paid), a-law, U-law, GSM, iLBC 20 and 30, Speex Narrow Voice Codecs: H264 (paid), VP8 Similar Projects: Zoiper SIP Webphone

    $500 (Avg Bid)
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    14 bids

    Need to design a load cell with 2 x strain gauges and 2 x accelerometers. The data will need to be recorded on the device itself for download and interpretation later ie. battery powered and recording data against a time signature. This will need to be built to follow a certain ASTM standard for the test requirements. These electronics will then need

    $1103 (Avg Bid)
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    8 bids

    ...simply want to convert my android phone to behave as sip voip gsm gateway like GOIP, DINSTAR. android phone will register with my softswitch via 3g internet and when my softswitch will send call to android it will forward to gsm network. so it will work like DIALER(DIALS NUMBER)--> MY SOFTSWITCH --> ANDROID PHONE --> TO NUMBER DIALED BY DIALER.

    $1273 (Avg Bid)
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    18 bids

    ...website comprising most of the items in this diagram attached: 'schematic minus BTS [url removed, login to view]' . The diagram should comprise a "MultiDSLA System; DUT UE; IMS; VPP+ (Reference SIP Client); DSLA (Analogue test interface); a cloud". similar to '[url removed, login to view]'. diagram should be in a style similar to the 3 examples given be...

    $60 (Avg Bid)
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    Animated AD for 60 sec. on charcoal range -- story on Acne , Black heads , skin allergy , underarm black ,dead cell removal-- five different theme-- with Voice over and Music , music will be provided by us . we need these five films in different cuts , 6 sec., 10 sec.,20 sec., 30 sec., and 60 sec., Animation film will remain same for all three AD x

    $411 (Avg Bid)
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    15 bids

    I need you to develop some software for me. I would like this software to be developed for Windows using .NET. Sample C# Project to :1. Connect to SIP trunk2. Make a call : play a file when pickup , hangup after playing , run event when hangup(by system or user) or no answer3. Receive call : play a file , run event when hang-up or no answer (by system

    $1049 (Avg Bid)
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    6 bids

    Hi, I'm trying to setup a Debian Jessie SIP server with Kamailio. Everything works but the TLS handshake doesn't complete. It stops at the client handshake, so the server doesn't send it's certificate. I would like someone who has experience setting up Kamailio with TLS and unix server administration. The deliverable would be to let me know

    $116 (Avg Bid)
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    3 bids

    Must be experience in Open Source IP-PBX development product. Setting up the infrastructure, build key features and testing. Works on Debian® 2.8 distribution, some of the key FOSS components that support the unified communication and management functionality are Asterisk®, FreePBX®, Chan_Dongle, and RaspAP Web GUI. Build a IP-PBX product that allows users to make VoIP calls to...

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    18 bids