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    8,420 flash media server sip робіт знайдено, ціни вказані в USD

    Basic router configuration (IP Address, hostname, access credentials)  SIP Trunk configuration from Cisco Voice Routers 1 and 2 to Genesys Virtual IP  ISDN PRI ports configuration  Dial Plan configuration  Inbound and Outbound Calls Testing

    $103 (Avg Bid)
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    16 заявки

    Hello, We need to develop a SIP to Viber, Whatsapp, WeChat and Facetime gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber, Whatsapp, WeChat and Facetime to complete the call to the called party number. The development platform/operating system is not important. The project should be completed

    $2757 (Avg Bid)
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    15 заявки

    ...tarjeta SD - Diagrama de conexión entre la raspberri pi y el telefonillo (speaker, mic, bell, DoorOpen) - Indicaciones necesarias sobre cómo añadir nuestra propia configuración SIP a la Raspberri....

    $198 (Avg Bid)
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    Sip Dialer With Include OpenVPN Закінчився left

    I need Sip Dialer With Include OpenVPN. Sip Dialer connect by OpenVPN. OpenVPN Connect only Sip dialer..... you need use PJSIP OpenSource SDK for SIP Dialer. I need Very Simple Apps... i need support g729 voice code. I can you OpenVPN server side. if you have good experience please contact me.

    $14 / hr (Avg Bid)
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    5 заявки

    We have A2billing server and we are building an app for our resellers to topup customers through app. We need the working API to Create and Manage customers - mainly Create customer with Callerid (including sip account) and topup.

    $161 (Avg Bid)
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    10 заявки

    download elastix pbx and connect it with sip and Asternic Call Center Stats Lite

    $27 / hr (Avg Bid)
    $27 / hr Сер. заявка
    5 заявки

    I need a website where I could sell mutual funds and insurances to my clients, and post regular information about the best funds and insurances. Also site must have SIP calculator and loan calculator. I need it preferably on wordpress.

    $125 (Avg Bid)
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    Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP

    $33 / hr (Avg Bid)
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    Sip.Net and Kamailio assistance Закінчився left

    I'm looking for programming help with [увійдіть, щоб побачити URL] and Kamailio. Must have experience with SIP and preferably experience with [увійдіть, щоб побачити URL] library. Will need 1 - 4 hours per week for 3 - 5 weeks.

    $13 / hr (Avg Bid)
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    Elastix 3.0 MT Закінчився left

    We would like you debug and fix our sip trunking /Ext to Ext calling setup to work with our providers so that we may use Elastix MT 3.0 to the fullest with multiple organizations and users per organization, Elastix MT.3.0 has been installed already just need to fix some bugs. We are currently able to have our handsets receive registration, and our

    $74 (Avg Bid)
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    My N...required job is as follows; 1- Design the telephony integration optimum system 2- Generate a block diagram and BOQ of required software and hardware (PBX, Virtual PBX, Handsets, Server...etc) 3- Configuration of the system 4- Handover Documentation including (Operation and Maintenance) Note: we will execute the procurement and the installation

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    ...(administrative network). There is a firewall that prevents connections from 4.0/24 to 5.0/24 We have VOIP SIP Server running at [увійдіть, щоб побачити URL] Currently the server runs well but we will like to make possible users on 4.0/24 to access the VOIP SIP Server If you have long experience working with this type of routers and you know how to make the ...

    $85 (Avg Bid)
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    KAMAILIO SIP + MEDIA PROXY WITH SIP CAPTURE INTEGRATION INSTALL INSTRUCTIONS + CONFIG FILES + TECHNICAL SUPPORT We need a KAMAILIO CONFIGURATION accepting calls from one NIC with public IP address and redirect passing the call to 3 different SIP providers connect to 3 different NIC’s with “local” IP’s (load balance - round robin). Media...

    $588 (Avg Bid)
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    Plataforma de Suporte Закінчився left

    ...informações de cartão de crédito. Analisaremos como será feito durante o desenvolvimento; Opcional (projetado somente para atualizações futuras): Ligar para o profissional (Chamada SIP, somente na segunda versão) *Funcionalidades – Aplicativo Profissional* Login; Logout; Checkin; Checkout; Envio de fotos de atendimento com...

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    I am looking for a Asterisk developer to quickly update existing Asterisk code for dialer, inbound/ outbound/ press 1, and other advanced features. Need to configu...Polls AMD (Answering Machine Detection) DNC ( Do Not Call) Support Inbound Campaign Custom Caller ID Multiple Trunk Support Realtime Campaign Management Multiple Technology (SIP) RESTAPIs

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    I need a SIP cordova plugin to use with Ionic 3. Cordova plugin is already exist. [увійдіть, щоб побачити URL] All you need is the create ionic plugin wrapper for cordova-plugin-sip.

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    FusionPBX CDR billing application Закінчився left

    ...are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers, the upstream CDR reports are not suitable for processing the clients billing for

    $200 (Avg Bid)
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    Asterisk Server rebuilt Закінчився left

    rebuild asterisk server using backup files sip and dialplan database AGI (python) restore all default functionalitys

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    ...Cisco 3945 router and it does not work fully. I have: Cisco 3945 with Voice license 1 x 4G Samsung phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over. Youll need:

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    TWILIO INTEGRATION Elastic SIP SETUP Закінчився left

    We have twilio and need someone to seup the Elastic SIP Trunking and make the phones dial out and ring... via Ciscso SPA540G and obihai

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    Resolve small bugs Закінчився left

    Hi guys, I have an android app which is already completed. It has few bugs on few phones. Can you resolve them? Are you aware of SIP protocol?

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    i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.

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    I need an Avaya IP Office expert who is familiar with SIP Trunk configuration with fax server products. Currently we have the licensing installed for 4 SIP Trunk Connections, but the Avaya IP Office is responding with a SIP "Temporary Error".

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    ...should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [увійдіть, щоб побачити URL]

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    RASPBERRY PI, Linux as a IP Phone Закінчився left

    I have third party Linux Server, SIP/VOIP, when an IP phones registered with server and are assigned with extension number like 501,502 etc, now these phones make communication with each other by dialing their number. What I want to do is, remove the IP phone and use Raspberry module, and it should work as phone, and connecting push button switch on

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    ...of work but getting frustrated with what looks like sip/nat issue and that I couldn't find a way to be able to call between local extensions. It's mainly for personal use, I wanted a simple PBX that could handle a small group of sip devices that could call each other and that could use an external SIP gateway. The aim was to have better sound quality

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    ...16.04 VPS server and then have all SIP traffic route over the VPN to allow 5060 traffic out of Dubai which blocks SIP on 5060. Install IPSEC on the Ubuntu server (currently in production hosting web app) Configure VPN Ensure only SIP traffic flows over VPN lock down the entire solution so the SIP traffic can only go to one IP address our SIP ser...

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    Trophy icon Sip and paint Закінчився left

    Paint and Sip for a Purpose. It's on September 13 2018 from 5 to 8 at the Palace Bar and Grill located 234 East Main Street in Waterbury. Tickets cost $25 includes all supplies and glass of sangria. Any questions contact Denise at 203-598 1781 dmartinez@[увійдіть, щоб побачити URL] Benefits NOI annual toy drive.....accepting donations

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    Гарантований
    $10
    3 робіт
    Calculator financial for website Закінчився left

    Make calculator for website SIP calculator, retirement calculator ,loan calculator

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    I have a, Asterisk, FreePBX and A2Billing Server installed in Google Cloud on a Free PBX. I need help to configured it so that I can bill departments in my I have already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers

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    3 заявки

    I have a, Asterisk, FreePBX and A2Billing Server installed in Google Cloud on a Free PBX. I need help to configured it so that I can bill departments in my I have already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers

    $122 (Avg Bid)
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    1 заявки

    I need 4 assemblies to be created in Profile Builder version 3: They are: SIP wall (including Internal and External Panels), Concrete Waffle Slab Floor, Mid-Floor and SIP roof. Each Assembly will be treated as a Milestone (the value will depend on difficulty). Outline of each assembly will be provided, but there is a lot of detailed information (both

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    ...purchase. Project Pitch We would need an experienced UX/UI Designer to create an appealing visual mockup for our upcoming new digital tool: the Social Impact Profile (SIP). The SIP will be a user database and part of our existing website ([увійдіть, щоб побачити URL]). With this tool we want to offer our customers the possibility to create their own user account

    $34 / hr (Avg Bid)
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    NaughtyTales Закінчився left

    I have written erotic stories for years. Some in which I have published and others in which I have written for other ...erotic stories for years. Some in which I have published and others in which I have written for other people. I love when I get in the groove and just let my mind wander while I sip my wine and free my mind... I'm sure you will too!

    $128 (Avg Bid)
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    VIBER WHATSAPP SIP GATEWAY Закінчився left

    Hello, We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Viber/whatsapp

    $666 (Avg Bid)
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    Appointment Setter / Cold Caller Закінчився left

    ...potential customers and setup sales appointments with upper management for website design and development. We will provide the call list of companies in our local area as well as a SIP line through Telzio to make the calls. The individual who wins this job must have good internet, at least 25/3Mbps to be able to make internet phone calls. Must possess a professional

    $11 / hr (Avg Bid)
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    20 заявки

    I have a Linux based SIP VOIP Server, by registering any SIP IP phone with this server and giving them unique number say 500,501,502.. and they works as VOIP calling system by dialing the telephone number. Now what I want to design is, instead of IP phone use Raspberry Pi as a phone and when registered with server it should work as a Telephone. for

    $431 (Avg Bid)
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    Project for DataHome -- 2 Закінчився left

    We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Viber/whatsapp executables

    $750 (Avg Bid)
    Угода про нерозголошення
    $750 Сер. заявка
    1 заявки
    Aplikacja VoIP Закінчився left

    ... Możliwość zestawienia kilku kont SIP od danego dostawcy. Możliwość wł./wył prezentacji numeru. Blokada połączeń przychodzących/wychodzących. Możliwość ustawienia prezentacji innym numerem. Możliwość rozpoznawania wybieranego numeru np.: krajowego lub międzynarodowego i nawiązanie połączenia dedykowanym kontem SIP. Możliwość zestawienia telekonferencji

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    Fix Asterisk + FreePBX + a2billing Закінчився left

    Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind NAT Problem: - Call receiver no ring - No audio as it should to tell balance, dial 9 for sip or iax Because this desktop behind NAT, i'll provide Teamviewer or Anydesk access

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    Trophy icon Design a Logo for "Virtuo" Закінчився left

    ...the manufacturer and vendor of the well-known carrier-grade Mediatrix gateways, is a global trustworthy partner supplying VoIP gateways for the Telecom industry focusing on SIP Trunking, PSTN/TDM replacements, Unified Communications, and Hosted Services. Its portfolio of VoIP gateways allows businesses to implement and manage reliable, robust, secure

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    Обраний Гарантований
    $91
    251 робіт

    I have an Esxi server The ESXI runs a virtual private server with Freepbx This is about home use What I need: Full system configuration: sip trunk, music on hold, extension, fax Ensure system integrity. Thanks, Shimon

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    Build an android based SIP Softphone Закінчився left

    Hi, I need an android based SIP softphone. We need a basic SIPphone only, so whatever standard festures are provided are ok. The program should be robust and free of bugs. So Ideally I will prefer somenbody who has build softphone previously. You can use any publically available libraries or kit. The UI ./ UX we will provide you.

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    ...Financial Planning ;Invest online;Testimonials ;Blogs; Careers ;Contact us With A live Chat options & watsapp messenger Few Algorithms need to be used –like calculator for – sip;retirement;marriage;child education etc...

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    I need a vb.net sip that connect to my sip server that dial a number and play a mp3 file. software must send my caller id i set to my sip server no dialpad needed

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    ...voice is like a lover who, at dusk, will curl around his lady beneath the shade of the moon who, at dawn, bears kisses to her long cream neck like a lover who will drink sip by sip from the river of her love Your voice is like heaven to a girl from the farthest green village who wears an anklet made from your coffee eyes who wears a sheila spun from

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    C & C++ EXPERT NEEDED FOR INTEGRATION Закінчився left

    ...HYLFAX AS AN OUTGOING ONLY BROADCAST FAX SERVER. Fax Broadcasting is sending an identical fax to multiple recipients 1. SOFTWARE MUST CONNECT TO T.38 SIP TRUNKS. 2. MUST BE ABLE TO SUPPORT 150 CHANNELS ON EACH SERVER. 3. MUST BE ABLE TO SEND 150 OUTBOUND FAXES ALL AT THE SAME TIME "SIMULTANEOUSLY" FROM EACH SERVER 4. ALL DRIVERS FOR THE SOFTWARE TO ADMINI...

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    Freepbx general support, particularly security, external sip phone connections like csipsimple not working, obi202 not registering, vpn connection setup, etc It seems there is NAT / firewall config issue between remote extension and freepbx. Access available via TeamViewer only and credentials will not be shared for security and privacy purpose. No

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    VB.net SIP phone to dial and play mp3 Закінчився left

    I need a vb.net sip that connect to my sip server that dial a number and play a mp3 file. software must send my caller id i set to my sip server no dialpad needed

    $100 (Avg Bid)
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    Cisco CUBE to Google Voice (Obiahi) Закінчився left

    Looking to integrate cisco CUBE with google voice /obihai device for google sip trunk for inbound / outbound calls. Create RP and dial plans on CUBE for trunk

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