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    8,268 sip client open source java 找到的工作,价格在 USD

    Sip client with the pjsip or other sip stack. Need the cloudy Address Book accroding to the API. Need the presence status. The project should be finished in 1 month.

    $1532 (Avg Bid)
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    VoIP Video IM等即时通讯功能。 支持Android,IPhone,Window Phone等客户端应用。 熟识SIP Webrtc XMPP。 提供Demo。

    $5555 (Avg Bid)
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    ...8、需要提供所有源代码和演示版本(演示版本的运行平台包括android、IOS),以及可交付的中转服务器; 9、并不需要完整的语音通讯软件,只需要能够实现语音传送、播放的功能模块即可。 10、演示内容:A用户点击“通话”按钮,中转服务器可通过设置好的IP地址,找到B用户,并发送通话请求。B用户接受请求后,双方可以进行通话。 11、你可以采用第三方组件,例如SIP,但是需要告诉我们是哪家的,效果如何。因为考虑到我国网络问题,所以建议是自己搭建服务器。 12、我们已经开发的软件可以在安卓、苹果商店找到,叫IEMAKER,通过安卓盒子,可以实现直播,但是目前还不能实现声音直播,因此,需要开发这个模块,用于该软件。

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    Cisco CP-7975G (SIP) not registering to CLOUD DEBIAN FREEPBX SERVER I DONT KNOW WHAT I NEED TO GET IT TO WORK. PLEASE HELP IF ANYONE KNOWS THIS SPECIFIC MODEL. THANKS.

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

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    ...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

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    Asterisk Server rebuilt 5 日 left
    已验证

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

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    ...migrated to a 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

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    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 1 日 left
    已验证

    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 [login to view URL]

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    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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    ...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 service This is a remote

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

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    ...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. I have already set up the following: 1. Free PBX & Extension 2. Free PBX routing and dial plan 3. Sip trunk to VoIP Provider 4. Outbound dialing rules 5

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    ...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. I have already set up the following: 1. Free PBX & Extension 2. Free PBX routing and dial plan 3. Sip trunk to VoIP Provider 4. Outbound dialing rules 5

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    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 ([login to view URL]). With this tool we want to offer our customers the possibility to create their own user account

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    NaughtyTales 已经结束 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!

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    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

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    ...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

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    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

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    Project for DataHome -- 2 已经结束 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)
    保密协议
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    Aplikacja VoIP 已经结束 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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    Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP beh...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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    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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    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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    I want to make and receive phone calls with my Twilio Number via ZOIPER. I understand most of the part, just have some problems with the Requesting VOICE URL. I think the job can be done in 20 minutes, i just need to see and understand how it works.

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    Need Twilio Sip Guide 已经结束 left

    I have a number on Twilio but dont know how to setup the sip correctly. I want to receive and make calls with a VOIP via ZOIPER5.

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    ...A developer should have a quality to take a mock-up of a website and bring it to life through coding. Should be proficient several different coding languages like HTML, PHP, Java etc. Additionally should have a vast knowledge of website platforms, front-end frameworks, plugins, and databases. • Advanced to expert coding skills • Strong understanding

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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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    ...ASTERISK EXPERT TO INTEGRATE 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

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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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    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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    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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    Portfolio Management 已经结束 left

    Following Function to be managed (Required With Source Code) 01. Purchase/Sales of Scripts from Market 02. Bonus/Split/Name Change of Company 03. Script Ledger Showing Stock and Value as per Book 04. Script Valuation as per Current Day. 05. Short Term/Long Term/Intraday Profit Loss Script wise/Datewise 06. SIP Management 07 Bank Payment/Receipt /Journal

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    hi I have flowroute as voip provider and using 3cx as phone system I want to set up texting [login to view URL]

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    We are VAS providers, We provide sms,voice and ussd services. Our servers are being hosted with AWS and we are us...using messaging,voice and ussd servers. We need the topology/blueprint of the total set up including [login to view URL] are connected with the MNOs with an SMPP connection for sms, Sip trunking for Voip. Complete set up deployment procedures.

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    I need help in setting up an OpenSIPS server and creating a SIP Proxy that alters some headers. I have knowledge of VoIP and SIP, but no experience in OpenSIPS. I'm a professional developer and do systems administration so i should be able to learn quickly. I would like someone who can teach me the basics, and prove me with answers to questions i

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    ...podstronie, np dedykowanej do dzwonienia, czy na każdej podstronie "należącej" do danego naszego konsultanta. To trzeba sprawdzić, tzn. czy jeden dialer podpięty do jednego konta SIP może wykonywać jednocześnie kilka połączeń. Kolejna sprawa to implementacja i integracja systemu płatności w sklepie (WooCommerce), np .PAYU, Przelewy24, itp. Prawdopodobnie

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    I have purchased a couple of numbers on Twilio. I want to use these numbers with a SIP client - such as X-Lite, Zoiper etc. I do not want the numbers to be forwarded to my cell phone or landline. If someone calls the number, I want X-Lite to ring. I also want to make calls from X-Lite with the Twilio number. I do not want to use a web app. Here is

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    Looking to hire someone who can get me a local Germany number on the daily basis with 10 extensions of SIP trunking. For more info, ping me.

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    Project for Rick B. 已经结束 left

    ...application that runs on Android phone and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. If android Layers have a problem for this job, you can use replicans OS. The application should then forward the audio/SMS and convert from SIP to GSM and vice versa. We can discuss any details over chat.

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    ...erotic writer to write an original story featuring a buxom femme-fatale woman who's big unwieldy breasts contain a milk that will shrink the size of any man who drinks it. One sip is enough to make him a midget and too much will send him down to bug size or smaller. Find a way to work this weird character into a sexy short story with a full plot and

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    ...device including mobile and landline phones We can use the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling protocol) with the multimedia framework and NAT traversal functionality. This API is suitable for all types

    $860 (Avg Bid)
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    Yeastar PBX Configuration 已经结束 left

    Require someone familiar with Yeastar MyPBX and S300 setup to configure for WebRTC & SIP

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