Voip sip mobile application工作

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    12,889 voip sip mobile application 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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    ...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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    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 expert for Voice API & SDK | VoIP Calling In Your Apps | Sinch and Java Script only.

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    I need a modulation algorithm, which can modulate any binary bit steam, to let the bit steam go through VOIP compression and decompression, and then be recovered with minimum distortion. VOIP for example is skype, or wechat, etc.

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    A mobile VoIP app solution to make calls from App to phone numbers all over the world

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [登录来查看链接] i want to run this program on a router Router Link [登录来查看链接] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    Programming Bandwith.com 4 天 left
    已验证

    Help programming sip endpoints, applications voice & sms in [登录来查看链接] I need to connect my [登录来查看链接] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [登录来查看链接] Prior experience

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    Multichannel Phone Dialer 4 天 left
    已验证

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    sip to Native dialer APP 3 天 left
    已验证

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many pe...

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    VOIP, SIP Expert 3 天 left
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    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

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    I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

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    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    SIP Proxy Server install 2 天 left
    已验证

    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Secure Issabel PBX (asterisk) 13 小时 left
    已验证

    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    Trophy icon Création logo Startup 12 小时 left

    ...startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip, Sip) Le nom du projet est "Caraib...

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    101 项参赛作品

    ...com/click-to-call/ etc You don’t even need a fax machine…,Receive Fax from customer as PDF in your Email. The system will support setting up/administrate the call center SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service

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    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    Siarum VOIP Phone System 已经结束 left

    We have developed a complete workflow management software the allows clients to use the CRM that is integrated with our VOIP phone system. It is built in asterix. We need developers to maintain and manage our asterix phone system within US timezone. This will be an ongoing project. Our website is [removed by Freelancer]

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    i need a custom voip switch like astpp which can handle above 1000CPS and CC with DID Management and recording with security

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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    voip portal 已经结束 left

    development of a end user portal for our wholesale VoIP business

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    Hi, my names Dan. Programs like Skype and Discord are great but I'm looking for a custom-made VOIP to use with friends and family, something that is more locked off. I will need the ability to make calls, group calls and or channels. I'd like to have a portal where I can manage members and they have the ability to also create an account. I hope I

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    Voip Android app. 已经结束 left

    I need voip app connected to my softswitch. if you have experience in voip app, please bid. Server is IPsmarx Breeze. Thanks.

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    Cisco voice gateway connection to SIP server and PSTN on E1 PRI

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    Need the below requirement Made for business Start with unlimited calling, unlimi...to set up and use Have your whole office up and running in a day or less. Respond easily to business fluctuations or customize your phone system using your online account. Mobile any time or all the time With an advanced CSM system where I can manage all subscribers.

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    sipxcom sip system setup 已经结束 left

    ...3.64, but i cannot setup outside domain, i want to use sip outside network, i know there are instructions but i cannot fix the problem. I know i just miss very little thing. i will give you teamviewer to setp up my pc. You need to do: 1)help me to set up sip system using sipxcom, i need the sip can be use in internal and external network. 2) tell me

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    I have just registered new firm in financial services. Stock broking, mutual fund, SIP investment advising, Various types loans & insurance services will be our products. I need logo & letterhead for this.

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    voip Solution 已经结束 left

    i'm looking voip app with desktop software and WebRTC Client

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    WE need to connect a cisco 7206VXR with VIC adapters for voice connection to the PSTN and to a Mizu SIP Server for users authentication and billing using sip/h323 protocol. We already done a major part of the setup and configuration of both the cisco and the softswitch but calls are still failing to authenticate on the cisco gateway to the pstn network

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    FreePBX is hosted on Vultr and voip phones are behind SonicWall. When there is only one phone is connected to FreePBX it works fine with no issues when we try to connect second phone from the same network it cannot register with PBX and can't receive any calls. Sometimes second phone gets registered and can dial out, but if we try to call some other

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    VOIP ASSISTANCE 已经结束 left

    VOIP ASSISTANCE SIP DEVELOPPEMENT

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    Trophy icon JT&T Brand 2 天 left

    I’m looking for a logo for business cards and letterhead I’m looking for some flyers so that I can advertise VoIP and SIP services the Name of the company is JT&T Communications we offer best in class SIP and VoIP. And phone system analysis design and implementation

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    566 项参赛作品

    I need you configure VoIP and internet settings.

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    1 - We need to configure Asterisk FREP...documentation from the work with detailed description with step by step of how to configurate it. 3 - The developer will register some of our SIP channels into the freepbx. We currently have both sip clients: - SIP Phone - SIP on PC 4 - We need someone who is an expert on it, because we need it asap. Thanks.

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    Project for Prostitov A. 已经结束 left

    ...discutere i dettagli tramite chat. A) TRUNKS INFOSTRADA Protocollo SIP username 390415541985 From Domain [登录来查看链接] secret GZD397XU Host [登录来查看链接] port 5060 Register String 390415541985@[登录来查看链接]:GZD397XU:390415541985@[登录来查看链接] B) TRUNKS MESSAGENET Protocollo SIP username 5406073184 From Domain [登录来查看链接] secret ajFFzpNj

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    Asesoria de proyecto VoIP 已经结束 left

    Proyecto: Implementacion de una mesa para practicas de laboratorio Voip con servidor Asterisk

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