Application android sip voip jobs

Filter

My recent searches
Filter by:
Budget
to
to
to
Skills
Languages
    Job State
    12,035 application android sip voip jobs found, pricing in SGD

    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

    $198 (Avg Bid)
    $198 Avg Bid
    3 bids

    Needing a VOIP softswitch for inbound and outbound calls. Need LCR. Was thinking opensips, CDR stats. I can build my own dashboard to query CDRs for billing.

    $3160 (Avg Bid)
    $3160 Avg Bid
    17 bids

    ...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 number.

    $925 (Avg Bid)
    $925 Avg Bid
    35 bids

    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

    $298 (Avg Bid)
    $298 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.

    $1563 (Avg Bid)
    $1563 Avg Bid
    11 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.

    $1789 (Avg Bid)
    $1789 Avg Bid
    18 bids

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

    $141 (Avg Bid)
    $141 Avg Bid
    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 should hear the audio file played. Please ...

    $664 (Avg Bid)
    $664 Avg Bid
    3 bids

    VOIP E Commerce API to create SHOP folder in JASON for wordpress -- 2 AWAITING ACCEPTANCE

    $857 (Avg Bid)
    $857 Avg Bid
    20 bids

    ...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 WebRTC projects is mandatory. Application to be developed

    $26 / hr (Avg Bid)
    $26 / 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.

    $207 (Avg Bid)
    $207 Avg Bid
    1 bids
    VOIP 4 days left

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

    $1377 (Avg Bid)
    $1377 Avg Bid
    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

    $28 / hr (Avg Bid)
    $28 / hr Avg Bid
    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

    $659 (Avg Bid)
    $659 Avg Bid
    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...

    $841 (Avg Bid)
    $841 Avg Bid
    25 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 DI...

    $1677 (Avg Bid)
    $1677 Avg Bid
    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...

    $79 (Avg Bid)
    $79 Avg Bid
    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

    $1382 (Avg Bid)
    $1382 Avg Bid
    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

    $153 (Avg Bid)
    $153 Avg Bid
    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....

    $1616 (Avg Bid)
    $1616 Avg Bid
    18 bids