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    7,901 java sip software mobile phone jobs found, pricing in USD

    We develop software for banks. When we demonstrate our software as part of the sales process, we like to have it look like a banks own application. We have 160 brands, and would like someone to look at their websites and capture their logo (in the best resolution they can find), and capture 5 colours: Their main brand colour, and text, the secondary

    $226 (Avg Bid)
    $226 Avg Bid
    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.

    $1192 (Avg Bid)
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    12 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.

    $1339 (Avg Bid)
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    19 bids

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

    $128 (Avg Bid)
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    8 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

    $487 (Avg Bid)
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    4 bids

    ...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). Min 1 year of experience in

    $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

    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

    $20 / hr (Avg Bid)
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    15 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

    $515 (Avg Bid)
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    ...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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    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...

    $1227 (Avg Bid)
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    19 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

    $20 / hr (Avg Bid)
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    1 bids

    It's simple I will rent a Linux Server (Centos u Ubuntu) and I n...(Centos u Ubuntu) and I need to install a dialler in it. The dialler is an Asterisk based one and all it's suppose to do is send call automatically to a telecom server by using a SIP account with 30 concurrent calls capabilities. The account has already been set in the Telecom Server

    $215 (Avg Bid)
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    16 bids
    ASTPP Fix errors Ended
    VERIFIED

    ...I am new on ASTPP. I use the link([url removed, login to view]) to manual install the ASTPP. I configure the trunks, rates, sip, gateways, etc. All are configured. When I try to make a call, show me error on fs_cli and the call hangup: 2018-01-13 13:00:25.568487 [ERR] switch_odbc.c:368 STATE: IM002

    $147 (Avg Bid)
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    13 bids
    ASTPP Configuration Ended
    VERIFIED

    ...create origination rate table and termination rate table 4. create trunk and set up routing so that origination carrier calls cam be terminated to termination end. 5. create a sip user account on new customer and route his calls to trunk of termination carrier. test and make sure all calls connect properly. give me a walk thru of the steps taken to

    $37 (Avg Bid)
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    2 bids

    ...Hosted PBX and Call Center. Based on Asterisk/Freepbx /Freeswitch etc. I would like to set it up in Amazon AWS so that we do not have to worry about the servers. Will have SIP trunking for the incoming and outgoing calls. Each customer will have their own portal to manage their users, extensions etc which can be done either with freepbx or if you

    $2174 (Avg Bid)
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    Buenos días, Buscando una persona para crear varios manuales/guias para la configuración de una cuenta SIP en centralitas Asterix y otras del mercado. Necesitamos una guia de configuración basica y con posibles errores - soluciones. Gracias

    $178 (Avg Bid)
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    12 bids