Sip simple asterisk presence工作

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    17,514 sip simple asterisk presence 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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    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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    主要指导elastix asterisk 的安装配置工作,熟悉Asterisk的模块(包括体系结构,配置文件,Log日志,History),能重新打包ISO文件

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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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    ...OVERVIEW I have to shift my existing asterisk vicidial server files ( my dial plans, modules , database ,IVR , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to a better hardware /server with Xeon processor 3.2 GHz.& 16GB RAM NEW VICIDIAL server details & its configuration ● Centos7, Asterisk, FreePBX & VTiger is already Installed

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    ...shall be awarded to individual who can install IP based PBX alongwith open source billing module ([登录来查看链接]). The IP PbX shall be asterisk based on linux platform. VPS for installation will be provided. The payment terms include 50% payout at the time of 50% work completion and rest at the project closure. Please

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    Looking for a cartoon character to use as a social media presence. Should be the size enough to be used as a profile picture. Idea: a gorilla (Athletic build) with kryptonian armor (with the Superman S symbol on the chest)

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    After login, receive message "cannot connect to asterisk"

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    Need help ongoing basis for insurance Restoration contractor

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    a2billing script -- 2 5 天 left
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    Dear all, I am seeking someone who can write a script that will install LAMP, asterisk, a2billing 2.1.1, and FreePBX automatically on CENTOS 7 by firing one or 2 command line(s).

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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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    I have requirement of asterisk PBX developer

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    ...Inventory/Asset Management Tools ([登录来查看链接] with GLPI) e. Visitor Management Tools ([登录来查看链接]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools (Jabber) Basically the idea is to help new IT Managers and IT administrators to manage IT infrastructure using opensource tools to

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    Hi -I want to install freepbx and configure termination gateways on my server using vpn. I already installed asterisk and a2Billing -configure a connection with clients

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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 5 天 left
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    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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    im looking for asterisk pjsip developer for support our system

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    Multichannel Phone Dialer 5 天 left
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    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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    Asterisk 16 Queue Call Back 4 天 left
    已验证

    We have 2 platforms: 1. Freepbx 2.11 and Asterisk 11 2. FreeBpx 14 and Asterisk 16 CallBack makes callbacks simple. When your agent receives a callback request, they will see all of the caller’s information in the screen. They can click one button to accept the callback request and [登录来查看链接] will automatically connect them to the caller. This process

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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 4 天 left
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    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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    need someone to fix the extension file on the asterisk script

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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 peppole

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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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    We have installed OpenAnswer, FreePBX, and Asterisk on a AWS. We are looking for someone familiar with the software to finish fully setting it up for us. This includes changing all the default passwords. After setting it up we want to upgrade to PHP 7.2, and NodeJS 8. We are also looking to remove all test files, and default configuration files that

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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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    we have a java script which is connected to asterisk we need someone to help change the settings

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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 setup an Asterisk Cloud deployment and need some help tweaking the final steps of the setup

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    ...portal, generate a voucher of the customers choice, enter the customers number, then click buy. The consumer will get a SMS containing Voucher number. (SMS sent out via local Asterisk Server) Major features for the initial design include: ● A minimal UI than allowed vendors to generate a unique token for client use. (UI needs to be made modern) ● App will

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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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    Asterisk Web development -- 2 1 天 left
    已验证

    Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc

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    Secure Issabel PBX (asterisk) 1 天 left
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    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 1 天 left

    ...companies. It's a 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 "Ca...

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    ...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 at any time, etc. Iam looking for a relationship with developer who will handle full deployment of asterisk with billing,GUI , with these functions

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    We are looking for a company who has 24/7 server administration. We have between 4-6 servers running at the same time, and we are looking for a new company, who can be in charge of everything concerning the servers. It is important that there is 24/7 support, and that there is system admins attached, if it is not only system admins. If every case goes through level 1 or level 2 support first, the...

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    I would like create agent survey on Issabel/Elastix. Need modification in UI as well like adding menuitem under call center reports section to show Agent survey results page within elastix/issabel UI.

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

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    we have 2 extension files we want someone to copy the dial out routine from one to the other

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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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    ... [root@pbx]# fwconsole restart Asterisk not currently running Running FreePBX shutdown... UCP Node Server is not running Running FreePBX startup... Starting Asterisk... [--------------->------------] 13 secs In [登录来查看链接] line 187: Unable to connect to Asterisk. Did it start?

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    ...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

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    ...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

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    We are looking for Java developer who can develop CTI & IM desktop software for our asterisk based PBX. for instant messaging we are currently using Openfire server but other XMPP servers are also welcome. We already have design and initial code to start with. Details of project will be shared on request.

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