Application android sip voip工作

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    13,020 application android sip voip 搜到的工作,价格货币为 USD

    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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    VOIP SWITCH 6 天 left
    已验证

    Need help configured my new voipswitch platform

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    WE WOULD LIKE SOME OF OUR IP PHONES TO BE PROGRAMMED. WE HAVE 15 MITEL 5224 AND SOME ,...OUR IP PHONES TO BE PROGRAMMED. WE HAVE 15 MITEL 5224 AND SOME , CISCO SPA 504G PHONES. THEY NEED TO BE PROGRAMMED. ALL TOGETHER WE HAVE 25 PHONES THAT NEEDS TO PROGRAMMED FOR VOIP. YOU MUST BE FAMILIAR WITH MITEL PHONES, IF NOT THEN PLEASE DON'T BID FOR THE WORK

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    We looking for an experienced Cisco Engineer to help us configure a Cisco 2800 series router to switch VoIP (SIP) traffic

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    IVR Product 5 天 left

    ...Maintain and manage Genesys Routing, Framework and reporting. Responsible for supporting call center routing strategies and have experience with Genesys reporting, URS routing, SIP and GVP technology. Good understanding of Genesys (CTI) infrastructure. Strong working knowledge of IRD and CME. A working knowledge of Genesys Systems Architecture to create

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    Need install asterisk and sip server on virtualbox for receive calls directly to computer and if 1st line is busy automatically forward to second free line with call recording and without monthly fee

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    Asterisk Queue Setup 4 天 left
    已验证

    ...+91-XXXXXXXXXX(Agent or user) come to my SIP and thus to my dialplan. If Agent calls he should be loged in as an agent If user calls he should be transferred to a Logged in user. Rewording the above requirement. I have my Agents say A1,A2,A3 on remote location. I have my callers C1,C2,C3,C4,C5 too on remote location. I have a SIP installed on my server which

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    We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone

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    I am looking for someone who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the required graphics and we also have an internet facing SIP server it will register to.

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    Hi altr, I'm looking for someone to help me r...using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure this SIP message is forwarded to the customer

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    Upgrading Cisco 7945G IP Phone to SIP Firmware

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    Find me some Leads for a VoIP Service. The Company offers cheap rates for VoIP calls via Mobile App (Andorid, Symbian, IPhone). Please, I do not need Email Addresses to send bulk Emails. I need some traffic twoards my site. And acctual leads that generating paying customer. Any suggestions appreciated.

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    Asterisck pbx server installation (fusion or frepbx), configure and integrate with Odoo VoIP module v10 for Web based calling using WebRTC protocol. Need proven or similar project experience in Odoo platform.

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    Voip - KAMAILIO- CRM 10 小时 left

    Mantenimiento y nuevos proyectos

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    voip control 9 小时 left

    voip control management and vpn

    $19 / hr (Avg Bid)
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    I want the page to be on Top of Google without Adverts. Linkbuilding, SEO Word setting, Forum postings, traffic leading, Facebook page etc.. Pls write suggestions how you going to make it no. one?

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    Develop a whatsapp equivalent mobile app using webrtc for voip and text + required backend.

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    ivr solution on java 已经结束 left

    I am providing vas on IVR by using java platform, so I need a java coder who as I need to get some changes done in coding , add new product and also create a SIP for OBD

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    VOIP Calling 已经结束 left

    I need a software that allows me to make phone calls on the internet from a free VOIP service. You must find the free VOIP service for me. I should be able to change the phone number from which the calls are made to random numbers each time a phone call is made. I prefer the code in VB.net but if you can not do it in VB.net please explain which coding

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    ...script will be provided but you will have to update it with notes. - Qualifications: I need someone that speaks great English, has organization skills, experience with VOIP (SIP) or has a similar program already so that we can check. Duties: Negotiations with the top officials of the companies; Sale of services in the field of B2B; Requirements:

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    Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload

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    ...only bid with in my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details

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    SIP to Whatsapp gateway 已经结束 left

    ...develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through 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/Android, or by using the Android mobile versions

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    Project for Tran H. 已经结束 left

    Hi Tran H., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I saw you were awarded to this project "SIP to Whatsapp gateway" did you finish it?

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    Want to build a google data studio dashboard displaying phone extension statuses, sip provider statuses, and outages for digium phone system.

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    Install VPN and setup a server connection Install OpenSIPS with graphical user interface install codecs connect GSM gat...connection Install OpenSIPS with graphical user interface install codecs connect GSM gateways to server setup OpenSIPS billing module install fail2ban and configure configure sip connections with clients and perform test calls

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    VOIP product 已经结束 left

    ...back end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically, we need to figure out what voip server and back end sip or trunk providers

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    Integrate Issabel-Asterisk with Twillio Elastic Sip Trunking to records Calls from Twillio (through Flex)

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    Skype Connect has the SIP trunk feature to use Skype as a SIP trunk of PBX. I tried to configure Skype connect with FreePBX but couldn`t make it work. If someone can do that and already configured such setup, I`m ready to pay.

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    ...from iso on a bluehost server. I can register my sip phone, I can make inbound and outbound calls. The only problem is there is one-way audio on the call. If I call from my Windows Sip phone to my cell phone -- My cell can hear me, but I cannot hear the cell. If the cell phone calls my Windows Sip phone -- The cell can hear me, but I cannot hear

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    Mobils apps for VOIP 已经结束 left

    I am a stand alone person who runs a FreePbx and A2billing platform I need android / IOS / and pc dialer that works via my platform.

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    I am looking for provider who provide me sip gateway for Indian operator but with open caller id and API. My concern is to send broadcast pre recorded voice.

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    SIP Proxy Server 已经结束 left

    Kamailio sip proxy server need with custom functions

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    JoyGSMDial 已经结束 left

    I need an application that runs on Android phones (4.0 or above) and can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. Please review and message me with considerations. If any requireme...

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    Asterisk PBX application to perform the following. To setup a SIP call between two servers using a specific codec. Have one side playback audio files. These servers will be connected over a WANem emulated link to induce packet loss/jitter/delay etc and produce degraded audio. Have the other record the degraded audio and store it .

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    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csipsimple to include the following features. - Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header

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    I am looking for Cisco VoIP Engineer who has outstanding knowledge on Cisco tools. Has to help me with projects/training. I am open with pricing part. Please feel free to reach out to me.

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    Dear all, We have a urgent Vicidial system to se...information and agent disposition data into one outlook. That means initial upload of list information must converge with output of agent disposition to create one report 6) To create sip trunk for testing 7) To create sample lead list 8) Testing and commissioning 9) Agent and supervisor training

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    SIP and CHEW 已经结束 left

    Its an cafe i want a logo for it

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    Memindahkan database dan script yang telah terinstall di server centos 7 yaitu menggunakan asterisk. Software telah tercustom akan dipindahkan ke server baru. Diharapakan yang telah memahami asterisk dan freepbx

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    SIP dialer for Android and IOS. Should support most Android and IOS versions including IOS version 12. Features: Runs behind NAT Support Bluetooth Echo Cancellation Push notification Call history display Account balance display Address book integration Support UDP and TCP protocols Support users to Login or Signup Supported languages: English, Arabic

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    1.I need someone to check my lan for see if it’s secure, and set me up firewalls ( pfsense ). [登录来查看链接] check cisco switch and routers [登录来查看链接] set up 2 voips phones 4. Set up windows servers

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    Need an Asterisk/FreePBX guru to show me how to setup xlite and Microsip sip phones to register with FreePBX extension.

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