Linux sip asterisk jobs

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    15,709 linux sip asterisk jobs found, pricing in PHP
    Configuring Asterisk 6 days left
    VERIFIED

    Configuring Asterisk to connect to my iPhone VOIP client

    ₱5264 (Avg Bid)
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    7 bids

    Set up an auto Fail-over trunk on asterisk, trunk will be given. Looking only for expert in asterisk to work with.

    ₱4935 (Avg Bid)
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    6 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)

    ₱981 / 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.

    ₱7947 (Avg Bid)
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    1 bids

    ...org/display/FOP/Installing+FreePBX+14+on+CentOS+7#InstallingFreePBX14onCentOS7-FirewalldBasicConfiguration I want Observium to continue operating at http://<hostname/ and asterisk admin panel to run at http://<hostname>/admin I expect this to be a very straight forward task. To be completed via teamviewer. Please express the amount of time (minutes)

    ₱684 (Avg Bid)
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    6 bids

    Hello, All! We have ASTERISK with realtime. We need develop RestAPI service that will extract peer information from ASTERISK 1.6 and send it to client. This service must have limits for some parameters like queries per second, answer with care. so it should affect on system performance. Write your way to do this in your bid, please. Or use chat

    ₱6024 (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

    ₱962 / hr (Avg Bid)
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    12 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

    ₱25513 (Avg Bid)
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    15 bids

    company name: Aleo Design a simple logo using cursive letter and a stylish handwriting font. font can be picked up from goo...cursive letter and a stylish handwriting font. font can be picked up from google fonts : [url removed, login to view] logo can have a small stylish star ( asterisk ) on the up right side close to the "o"

    ₱9618 (Avg Bid)
    Guaranteed
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    336 entries

    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

    ₱63731 (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...

    ₱3037 (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

    ₱53100 (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

    ₱5872 (Avg Bid)
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    3 bids

    Cannot connect to Asterisk error on freepbx -need freelancer to assist in fixing

    ₱962 / hr (Avg Bid)
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    6 bids

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

    ₱57960 (Avg Bid)
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    16 bids
    Project for Santosh S. 7 days left
    VERIFIED

    Hi Santosh S., I noticed your profile and would like to offer you my project. Im getting a error on freepbx that asterisk not started and need assistance urgently if possible

    ₱557 / hr (Avg Bid)
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    3 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

    ₱1012 / hr (Avg Bid)
    ₱1012 / hr Avg Bid
    1 bids
    CONNECTED DONGLE LTE HUAWEI 4 days left
    VERIFIED

    I NEED CONNECTED MODEM 4G HUAWEI for Asterisk AND USE IT IN VOICE

    ₱7694 (Avg Bid)
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    7 bids

    It's simple I will rent a Linux Server (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

    ₱10985 (Avg Bid)
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    16 bids
    ASTPP Fix errors 3 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

    ₱7441 (Avg Bid)
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    13 bids