Windows mobile voip sip jobs

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    12,030 windows mobile voip sip jobs found, pricing in ZAR

    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)

    R238 / hr (Avg Bid)
    R238 / hr Avg Bid
    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.

    R1927 (Avg Bid)
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    1 bids
    VOIP 6 days left

    Cisco Ip phone configuration,implementation,imprensence,expressway c and e,unity server.

    R12824 (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

    R245 / hr (Avg Bid)
    R245 / hr Avg Bid
    11 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

    R6406 (Avg Bid)
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    14 bids

    i need some one that can installe the API of a VOIP Ecommerce here is the files [url removed, login to view],i also need SEO campagne in USA, Canada, Europe, i have a budget of only 100 euro per month and i can get it on to 12 month, if your not interesterested to be payd monthly please don't answer this project, thank you...

    R7965 (Avg Bid)
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    22 bids

    i 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

    R15451 (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...

    R736 (Avg Bid)
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    12 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

    R12874 (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

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    3 bids

    ...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 each other and give access to a long list of features....

    R14052 (Avg Bid)
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    16 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

    R245 / hr (Avg Bid)
    R245 / hr Avg Bid
    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

    R2823 (Avg Bid)
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    15 bids
    ASTPP Fix errors 4 days left
    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

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    13 bids
    ASTPP Configuration 3 days left
    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

    R454 (Avg Bid)
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    ...Web site, blogs, articles, copywriting, social media writing, translation, proofreading, press releases, brochures. Desired Candidate Profile: Must have good knowledge in VOIP Technology. Excellent overall writing skills in a number of different styles/tones. Develop unique, error- free, grammatically correct content for websites. Edit and revise

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    10 bids

    iVoipe is a telecom company based in ...telecom company based in US , New York. It is connected to more than 250 providers and offer A to Z termination with very good prices and quality. We are recruiting for a VoIP reseller who has knowledge about selling voice calls to people and experience in Itel switch platform. (no other requirements needed)

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    Voip Trader Ended
    VERIFIED

    revision and new graphic layout of an existing presenation

    R6811 (Avg Bid)
    Featured Urgent
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    1 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

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