Sip simple asterisk工作

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    16,713 sip simple asterisk 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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    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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    URL Free to Call System 6 天 left
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    URL Free to Call System from the website, to use my sip trunk for clients to call me free.

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    ...DTMF button on screen so user can press options from IVR and user use inbuilt microphone then connect call to SIP address on Incredible PBX OR When number clicked on "Free Call" button, viewer types in his/her phone number and submits to SIP address which setup on Incredible PBX Example [登录来查看链接] The right candidate will be also contracted

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    ccnp voice 6 天 left

    ccnp voice training and configuration for voice gateway mgcl sip and h323

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    A2billing Installation manual 6 天 left
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    ...to have another server with A2Billing installed on it, so I need someone to write the installation guide commands to establish it completely on CENTOS 6.7 VPS starting with Asterisk, a2billing & freePBX , I will be using it for PC2Phone AND calling card, so the installation manual you write should indicate the proper setup. I know that there are alot

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    Hello we need a person that can setup the ASTERISK / FREEPBX server for us + make manual how to use it. Including all security + usability settings. Budget $100. Must be done in one day.

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    I need an app for Android and IOS that basically takes and makes sip calls. It can be an open source phone labeled with the name that I need.

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    ...Centrales telefonica Asterisk y GrandStream (con acceso al personal de soporte directo del fabricante). -Configuraciones de troncales SIP dinamico y estatico. -Facilidades de troncales SIP con las prestadoras de República Dominicana. -Configuraciones de IVR. -Cola de llamadas. -Extensiones SIP. -Configuraciones de PBX redundante Asterisk -Configuracione...

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    Fix bug and implement call in asterisk application using ari. The application is built on node ari package

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    Asterisk has stopped running on my version of Freepbx. I do not know how to access the command line or the commands to restart. I just need someone to show me, preferably by doing a screen share and walking me through the steps on how to 1. access the command line on my server 2. enter the commands to stop, and restart asterisk

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    centralita 4 天 left

    Tengo un gateway goip gsm de 8 sim, 4 voipeadores tplink y un servidor en asterisk pero necesito crear la troncal sip y enlazarlos con los voipeadores para realizar llamadas salientes por el gateway gsm

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    I want to setup click2Dial function on my PHP website with Asterisk, the open source communications framework. There should be call button on press call should be the dial. and store the recording of that call. Have to setup Asterisk on AWS.

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    A2billing installation 4 天 left
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    I need to install a2billing in freepbx 13 with asterisk 11 and have to verified the command script by using teamviewer in my virtual local server. Kindly also need to show some configuration, like in freepbx IVR has two option and one of them is calling card option. so if customer chose the calling card option then how to divert them to a2billing and

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    Need a SIP endpoint built in Twilio to point traffic to my freepbx installation. Preferable if Twilio can only send 10 digits.

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    Install & Setup Free PBX on Digital Ocean Droplet using Centos Setup with carrier, all basic features etc. I am new to this so will need walk thru after all is setup.

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    ...M., I want to install a new asterisk server. We will use it as a SIP proxy. We will have a blacklist B numbers in db (or anywhere), we will sync this data every minute.
And Asterisk will check if the incoming all is in blacklist, and will route the call after. The flow would be like: call comes from A server, to Asterisk server with 3333 prefix 3333XXXXXX

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    Need an ICE/STUN/T...have NAT WebRTC clients audio working fine with my Asterisk. Need to check and explain me how to configure Asterisk and WebRTC script (like doubango) to work when the client is behind NAT. I have coturn installed but not configurated. I see RTP packets but only in one way. Have server A with Asterisk and server B to WebRTC script.

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    Hi Toshazed, I need your help to install a2billing in freepbx 13 with asterisk 11 and need to show me the command script. it will be install in my virtual local server. Kindly please help me to show other configuration which will be discuss after installation.

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    Conexion CRM Bitrix e Issabel 2 天 left
    已验证

    ...conectar nuestra Central telefónica Issabel (Asterisk) con el CRM Bitrix 24. Lo que queremos lograr es poder realizar llamadas telefónicas desde el mismo Bitrix a través de las centrales Issabel ya configuradas. El bitrix cuenta con instructivo para conectarlo con FreePBX. Si bien este se maneja también con Asterisk, es necesario hacer la configuraci&oa...

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    I need some help to fix up some call problem with an Asterisk server - All internal are behind NAT (with Miktotik router) - Asterisk server are on a VPS on cloud

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    Telegram to SIP Calling 2 天 left
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    HI We need to have a SIP trunk between a Telegram ID ( [登录来查看链接] ) and a SIP server, so every body who calls our telegram ID using Telegram messenger, will be forwarder to our SIP trunk,

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    We are looking for a core PHP expert with experience in OOPs, MVC, 3 tire architecture, video & audio chat using asterisk and e-commerce for online training purpose.

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    ...We have a current very old custom build solution based on a Asterisk PBX with ISDN and want to move it to the cloud and provide new features for our team. Its very important that you have indept knowledge arround Asterisk PBX and integration with SQL / External scripts from within Asterisk PBX We can accept that you just provide the SQL and we can

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    ...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

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    ...text content and form field content. Since we are converting the content from the web to print we do not need to include "* required information" and don't need to use the asterisk * On page 1 of the PDF we don't need to include the text so just disregard the following "Instructions to Applicant 1. You must fully and accurately complete the Application

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    Need someone to remotely help me with configuring an IPPbx solution. Consists of: 1 HA100 2 UCM2510 100 extensions 4 conference rooms Single-level IVR configuration 2 SIP trunks configuration Write a manual to start a conference room Write a manual to add an extension Write a manual to make a backup I need someone to be able to provide remote support

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    We need a server specialist that can install tonight (with information from our developer) an Asterisk VoIP Server on a CentOS. Next to installation and proper working of software we need you also to increase the security settings of the server. Budget $100. Must be done in one day.

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    Fix small issue with Goautodial 22 小时 left
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    Looking for someone who are expert in fixing the asterisk server and goautodial server to fix some issues. I am trying to connect that server to external public IP behind the firewall/router. Somehow there are some errors such as " handle_request_register: Registration from '"1300" <sip:1300@[登录来查看链接]>' failed for '[登录来查看链接]' - Wrong password

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    I have a Asteris 11.14.1 PBX. I want to use Bitrix24 CRM and log all incoming and outgoing calls in the CRM. I need to connerct Asterisk with Bitrix24 for inbound and uotbound calls

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    I will require a software to send and receive sms sms messages through sip protocol or sip simple. It will feature loading of contacts via CSV file, inbound csv report file exportation, outbound CSV report file exportation, capability to receive sms and sender sms messages, contact management, ability to send to 100 contacts at the same time or more

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    I am setting webrtc to an Asterisk server,I need an engineer with experience in networks and Asterisk webrtc configuration, in order to fix error net::ERR_CERT_COMMON_NAME_INVALID. I have a Goddady certification. Asterisk show TLS/SSL certificate ok.

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    Asterisk Call API 5 小时 left

    ...request this web address, Asterisk will: 1-) Call {CALL} via {CALLERID}. 2-) After CALL answers, send call status (busy, rejected, answered etc), duration and recorded soundfile path to our REST API service. - Need support over 50 calls at same time. - I have a server at digital ocean. Asterisk is not installed. - I have a SIP account allows caller

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    VoIP / SIP Softphone 已经结束 left
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    We are looking for a developer that has preferably already developed a working softphone that can be customized with our logos, colors and text. Phase 1 will be developed for windows, phase 2 will be for Android and iPhone. Alternatively, we are willing to work with a developer to create the software from scratch as long as the delivery timeline is acceptable. Our expectation is that the softwar...

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    Sip Audio Intercom 已经结束 left

    I want to use a rasberry as a sip intercom ,I want to dail the Pi and listen in via the mic and press *46# on my phone to triggger a output dtfm coding will be the way.

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    ...if I try to solve the pages does not update correctly the Vicidial/Asterisk MySQL tables. I need a correctly full working installation, the result should be an installable and fully working ISO or Virtual Machine harddisk with the working installation. Requires: dkms-dahdi, asterisk, vicidial, php7, MySQL, MariaDB, CentOS, and other of the same kind

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    asterNET modify sip trunk 已经结束 left

    asterNET modify sip trunk need a vb.net or c# to modify sip trunk in asterisk from a windows base software

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    Calling using a callcentric sip trunk provider cannot listen anything from another side.

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    Develop a Sip client 已经结束 left

    Hi i need a Sip client similar to Linphone Android. You should use the [登录来查看链接] for the project.

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    Flexisip server setup 已经结束 left

    Hi i need a Flexisip server to be set up, complete in all respect, with all configuration for working with my Android sip client for Voice and Video communications.

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    I need Android sip/voip client to my sip server . if you have experience in this voip/sip app, please bid. I will provide sip details of my server. Thanks.

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    A simple code to play am mp3 file or a wav file over a SIP call or VOIP call. I have a VOIP server. I used Ozeki SDK before buts it not free. So I want a simple code to do exactly what Ozeki SDK will do.

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    Hello All, I'm looking an experienced Microsoft system admin. The freel...windows server 2016 needs prior to be prepared before the installation of Exchange server 2016. UC must be configured as well since we will also be hosting VoiceMail for an Asterisk VoIP system Note: Apply on this job only if you already perform such as job. Regards, Léon

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    ...windows. It works fine. However its flakey I have a Cisco 3945 with Voice/UCS license that I'd like to run a SIP server on IOS to replace this. I only use this for LAN, there is no PSTN calling. I need my LAN devices to register to the SIP server and also my 4G Mobile device (using Zoipher client) when it's not at home (it uses a Split DNS entry to

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    ...of the APP: 1) voicemail is forwarded to an external number. 2) the forwarded voicemail number is an external CLOUD / VOIP external programming phone system like Twilio, or SIP or you tell me what you recommend. 3) Call comes in, all calls not on contact list or "white list" are Declined and sent to voicemail. 4) call goes to voicemail (the VOIP),

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    I am looking to build a small application to be able to receive H.323/SIP video conference calls to internal IPs. The application would then receive the video and slides that the caller shares from his Polycom / Cisco codec and process it as an H.264/265 video stream into a streaming server like WOWZA / YT / FB etc. Preferably in .NET technologies

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