Application android sip voip工作

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    12,964 application android sip voip 搜到的工作,价格货币为 HKD

    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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    voip语音网关模块 已经结束 left

    具备sip协议,支持30路以上fxs,可以实现点对点通信,需要提供相关技术支持

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    直拨,回拨,点对点语音,视频电话,文字聊天,可见好友位置,语音对讲,文件传输,充值支付,一键语音客服,邀请好友。业务管理系统。(Removed by Freelancer.com admin for breaching Terms and Conditions)

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    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.

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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。

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    1、开发一个用于移动APP软件的即时语音通讯模块,发送端是IOS或者android平台,接收端是android平台; 2、即时通讯模块功能主要实现单对单语音即时聊天,无须实现视频聊天; 3、要求操作简单,点击“通话”功能按钮即可实现通讯; 4、由于即时通讯双方可能位置很远,因此必须经过互联网传输,需要中转服务器进行语音媒体的中转。中转服务器可以采用成熟的第三方软件或者协议,但是在项目完成后可以交付给我们; 5、发送方网络的最低可能是ADSL 2M。但是延时不能超过1S; 6、语音质量要高,不能有杂音、回音等问题,需要可以用于商业行为的那种; 7、需要有可供调用的组件和调用方法,我公司可通过调用方法进行上层功能的开发; 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...

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    i need voip website wuth all tools to start here in kuwait jjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjj

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    voip website 已经结束 left

    i need voip website wuth all tools to start here in kuwait jjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjjj

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    I need physical Turkish phone numbers for Microsoft Azure verifications. I can buy up to 20 phones per month but all of them must be non-voip and didn't use on Azure before.

    $16 - $63 / hr
    $16 - $63 / hr
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    Rebrand Chamilo or Moodle elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of sl...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops.

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    We are looking for a VOIP expert to help us to figure out away to make unlimited phone calls to the United States and Canada for free. With NO volume restriction. We know its possible if you know of a way. We would like to hire you as a consulting. The quality and reliability must be high. If you know how to do a proxy server over google voice, skype

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    Hello! I am looking for a name and logo for my new painting party business. I offer mobile(no brick and mortar studio) paint and sip style parties mostly at local bars/restaurants and also private in home, but do not want to be limited to just canvas painting as I would like to host other craft classes as well in the future.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Hey Need to add a new feature ASTPP VOIP BILLING ASTPP use freeswitch pbx, but he generate diaplan. wee need add function pitch shift

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    Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    AXNET VoIP 3 天 left

    ...someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones like cisco, Unifi VoIP phone and Polycom. Thanks

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    Hello, We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough

    $259 / hr (Avg Bid)
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    Trophy icon Study Cafe Name is needed!! 6 天 left

    NAME IS NEEDED. NOT A LOGO!!! I NEED A NAME SUGGESTION Study Cafe Name needed! - This is a space where you can study and get sip of coffee and some desserts. - space is mainly for studying (and maybe working) - there are space for them to do team work - there are space for customer to rest too - please see some example pictures.

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    45 项参赛作品

    I need to integrate incoming call and outgoing call in my website by using SIP library. Website must be developed by using ASP.NET MVC. I need an expert guy. I have sip server details. Only expert guy apply please. Open suggestion for js library but must support of my server details support.

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    We are looking for an artist to create desirable pieces that the general public would like to sign up and...instructions to replicate which we can share with our local artists to replicate in a classroom setting (photo+text). Hoping for quality work very similar to the top paint and sip companies out there, and would be willing to work with long term.

    $235 (Avg Bid)
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    1 项参赛作品

    ...an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance with the ITU-T SIP-I specification Q.1912.5. - The SIP signalling shall transport a 180 RINGING message without SDP parameters meaning the originator should

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    ...API 2. Call is initiated via Twilio Outbound Voice API using a Phone Number. 3. When call is answered, the call is forwarded to another Twilio phone number that uses Elastic SIP Trunk. It is attached to our IBM Voice Agent Service. 4.. Once Voice Agent is connected the user will speak to it., then be transferred to another phone number. We are looking

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    VOIP Apps for android and IOS 2 天 left
    已验证

    I need a developer who can develop VOIP apps for ios and android. App features, look & feel and functionalities will be very similar to rebtel apps. [登录来查看链接] I will provide design files.

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    I wanna run voice campaign so i need a solution from where i am able to run voice campaign.

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    We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complete system: BigBlueButton supports ...configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    VoIP Termination Expert 1 天 left
    已验证

    We are a company specialized in termination. We have many destinations with several operators to handle , each has his own unique system. To keep track of our business and to be able to make sure everything is working properly, we need an expert in cloud configuration to manage the cloud configuration. We know exactly what settings we want to apply, all you have to do is the proper configurations....

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    I am a asterisk Expert who can easily understand about auto dialers, i want to make a web application that allows autodialing using voip

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    kamailio5 SIP confing with RTPproxy 15 小时 left
    已验证

    hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [登录来查看链接] if you can make it please bid

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    Need to develop SBO VPN to compress VoIP bandwith,

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    Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5

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    I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.

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    ...to send call server to local voip gateway like server in USA VoIP gateway in Bangladesh server to Bangladesh Openwrt Router connected BY VPN and VoIP gateway connected to openwrt router so i send call from server its work fine but i need some one who can help me to create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but i want

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    if you developed any Android Voip app and have source code, please bid. I will discuss about it in details and let you know what i need it. Thanks.

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    I have a bunch of phones ...firmware update when they are factory reset. I have a DHCP 160 option pointing to HTTPS server for provisioning. Firmware is also on that HTTPS server but my phones connect to my SIP server correctly and ignore the firmware update task. I need help getting my phones to force firmware upgrade when they come up and provision.

    $78 - $235
    $78 - $235
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    We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand

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    ...[登录来查看链接] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight) 10) Spread Order Entry Response (Overnight) 11) Switch Order Entry 12) Switch Order

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    Android Voip APP. 已经结束 left

    if you have Android Voip app source code, please bid. I will discuss details . Thanks.

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    I need to add call history and contacts function.

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    I need 10 minutes help (remote assistance /Teamviewer) to resolve an issue with Miktotik Queue with mangle for VOIP

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    I need help to configure Mikrotik queue for voip traffic sharing my desktop in teamviewer.

    $72 - $271
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    Application key features: * high definition audio and video calls * audio conference calls with various participants * pictures and file sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers

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    Homer Seven SIP Capture 已经结束 left

    I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP

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    FreePBX Setup 已经结束 left

    We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your

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    voip callshop setup -- 2 已经结束 left

    Hi There, Looking someone in order setup A2billing + Callshop module, calling card, and ivr setup. Thanks to reply if you got some experience on it

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    I need for someone to build a simple 3 page wordpress site and for this person to have completed numerous twilio api jobs.

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    I need someone available to build ios app. This app will have chatting(basic feature: text/video in group), VoIP, whiteboard, payment api integration. Please don't hesitate to share your skills and experiences.

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    Hi We are looking for freelancer who is aware of complete integration and workflow of GrandStream HT813 with Issabel. Issues to resolve Call Progress Dial Tones to be set for Airtel (India) Call disconnection (FXO Termination) to be set Other settings Secure Connection between device and server Complete Trunk related settings in Grand Stream and Issabel (Elastix) Only bid if you have complete ...

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    ...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m

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