Android sip client asterisk pbx jobs

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    18,270 android sip client asterisk pbx công việc được tìm thấy, giá USD
    Project for Vu L. Đã kết thúc left

    Hi vule83, mình đang có dự án Odoo CRM tích hợp tổng đài voIP sử dụng Asterisk 11. Bạn đã từng làm dự án nào tương tự chưa? Nội dung cụ thể thì cho mình số dt của bạn để trao đổi cho nhanh nhé

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    Chào các bạn, hiện tại tôi đang rất muốn kết nối vtiger opensource 7 với tổng đài assterisk, nhưng tôi không biết làm việc này, rất mong các bạn hỗ trợ tôi kết nối giữa 2 ứng dụng này, tôi sẽ trả phí kết nối cho các bạn. Trân trọng cảm ơn!

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    Project for nttranbao Đã kết thúc left

    Hi nttranbao, I'm a newbee asterisk. Now, my project are doing function monitor video call with asterisk. Do you do that function? If you can, you can send email for me ( [login to view URL]@[login to view URL]). Thank you :) Hy vọng anh biết tiếng Việt. vì tiếng anh của em không tốt lắm. Em mới nghiên cứu asterisk và hiện tại dự án của...

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    Mình cần hỗ trợ config Elastix, inbound, outbound và sip trunk

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    change logo elastix PBX Đã kết thúc left

    Tôi cần thay đổi logo tổng đài elastix I need Change logo elastix PBX

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    Twilio SIP set-up 6 ngày left
    ĐÃ XÁC THỰC

    Please set-up Twilio phone number with SIP so I can use on following site: [login to view URL]

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    We are a Philippines Company looking for an expert in Docker, Kubernetes and Google Cloud to help us speed up migrating 3 existing VMs (Java App, Asterisk and Postgres) to Docker Containers deployed on Kubernetes on Google Cloud. We have already containerised the Apps. We now want to accelerate deployment and to then have assistance with "peeling out"

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    I need a complex asterisk job ... need experts only.

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    I'm looking for a tech who already has completed a Ringless Voicemail drop system. We are US based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid

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    VOIP - CALL DROPS 5 ngày left
    ĐÃ XÁC THỰC

    We use a XORCOM ELASTIX VOIP and have 2 office locations with a VPN tunnel. The second office cant make or receive calls which seems to be a SIP ALG or NAT issue. But need this fixed.

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    ...instances, PHX-PBX and BDQ-PBX. PHX-PBX is my main PBX, and has most of my configuration (e.g. Extensions, IVR, MeetMe, etc.). On BDQ-PBX, however, I have a Sangoma A200 card and a single PSTN connection on Port 1. What I need to do is the following: 1. For all inbound calls to the PSTN line, I’d like them routed from BDQ-PBX to PHX-PBX...

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    OBD CALLING 4 ngày left

    Hello We have SIP,PRI and want to build script where we can make OBD Calling (asterisk),

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    ...DESCRIPTION I need a video clip to start a tutorial video on Centralino Agile. Centralino Agile is the cloud PBX offered by TED that brings Unified Communications to Small Business Enterprises. This clip should give the idea of an Agile PBX (this is the meaning of Centralino Agile) with a nice logo and its animation. An idea may be: - a phone handset

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    We want to use our freepbx as a call center solution. We need to develop an auto dialer module similar to Vicidial for our freepbx system. FreePBX is a great PBX system. Now we need to make it a complete package for telemarketing. Most of the ViciDialer should be there. Regards, ALSOFT

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    ...passing correct caller id to my PBX. Caller ID has two parts. One is Caller ID name and another is Caller ID number. Caller ID name part is okay and gateway sends is as the actual caller ID. Where is Caller ID number is incorrect. In Caller ID number gateway sends the DID number, through which it is connected with the PBX. Please see the attached image

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    OpenVOX GSM gateway configuration changes. 4 ngày left
    ĐÃ XÁC THỰC

    ...passing correct caller id to my PBX. Caller ID has two parts. One is Caller ID name and another is Caller ID number. Caller ID name part is okay and gateway sends is as the actual caller ID. Where is Caller ID number is incorrect. In Caller ID number gateway sends the DID number, through which it is connected with the PBX. Please see the attached image

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    ...checkout page the fields, Pais, Endreço. Cidade,Estado, CEP, they are [login to view URL] I do not know why the red asterisk does not appear. I installed the plugin WooCommerce Checkout Field Editor, although I enter the fields as mandatory, the red asterisk does not appear , and it is not possible cancel the writing (OPTIONAL). mysite: [login to view URL]

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    Looking for an experienced contractor to update configuration on our VOIP environment (a dozen or so phones). Currently using PiaF / FreePBX but happy to change to another distro if needed. Most phone are Mitel/Aastra 6739i and plus a couple cheaper aastra and a polycom conf phone. Key Issues / Targets: 1. Hot-desking - Configuration to allow for user speeddials to move with users as they log i...

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    Build a logo 3 ngày left

    Need a logo for pickleball paddle Company. Need to be PBX I want the "X" to be in that shape. Should be sophisticated and minimalistic. Darker colors would be preferred. Thinking out of the box would also be great

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    ...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed

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    Integrate PBX ( Cisco, Avaya ) with Middleware for Phone Lock/Unlock, Assign Name to extension, create Voice Mail, Set/Delete Wakeup Call etc. More info will be shared when the candidate is selected.

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    Looking for someone who can configure Dinstar GSM Gateway to work with my local asterisk setup.

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    Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.

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    Hello, We need to add a functionality to our IVR which is based on Asterisk V 13.14.0 / PhpAGI. Os is Debian 8, database is MySql 5, Php is also 5. Simple functionality: - Inbound call accepted (client who needs support) - IVR (PhpAGI) says "welcome" - Call is forwarded to 1st level agent (already done by DIAL command) - 1st level agent takes call

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    We have a software appliance (PBX) which we want to deploy on AWS multiple times, via an AMI. The source needs to be compiled onto a suitable Debian OS, tested to confirm it runs, and then an AMI needs to be created from this in AWS. Appliance source code: [login to view URL]

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    ...like to offer you my project. I need a costume softphone for Android and Iphone using Belledonne Communications SDK or other softphone SDK. Basic requirement 1) Upon installation of Softphone, prompt for username and password 2) It will then login to a website (MySQL) and get the pbx server hostname, username(ext), password, port (5060 or 5160) and

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    ...offer: - hourly wage: 15 USD/hour; - wages minimum 10000 UAH per month; - work in a prospective company: we introduce automation systems based on open-source products Asterisk IP-PBX and CRM VTiger: open API, easy integration with other systems, more than 30 own developments for CRM VTiger, VTiger has wikipedia and community community developers; -

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    ...(landline 1) and I have my own server too. I'd like to redirect calls made to landline 1 to another landline (landline 2) that I don't own. I've already made a vocal robot on Asterisk so that depending on the digit the caller presses, it does something different. What I need now is the last part : 1) Depending on the digit pressed, forward the call to a

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    A call center with yearly more than 5 million calls wants to optimize the customer experience by reducing customer waiting time and drive down operating cost by using "Virtual IVR" technology. Within three month you need to implement the software. Use-case: customer calling can choose to receive an SMS to his phone to access the Visual IVR system for self help. The SMS contains a link th...

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    We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization

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    App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.

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    CONVERT OUR SITE TO WORDPRESS Đã kết thúc left

    ...including MENU and FOOTER. Except for this it could be a plain HTML site. It has good SEO position on several search terms in Portuguese to our marketing. Our marketing is IP PBX, Automated Dialer, IVR, Chats and Chatbots. Our goal in such conversion is to improve maintenance and content addition altogether with: SEO IMPROVEMENTS ADWORDS COST REDUCTION

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    ...require someone hands on experience in installing goautodial or vicidial solution on Google cloud computing. Good understanding of Linux, Asterisk and Vicidial is essential. We wish to start testing Asterisk and Vicidial in the cloud from Google. We require setup, testing and ongoing support. Phase one is setup so we can start testing. Phase 2 will

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    ...outsource for those tasks when needed. Currently i can be specific on a project which you can tell me you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry

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    Provisioning cloud Cisco 7911 Đã kết thúc left

    Objetivo: Provisionar teléfonos Cisco 7911G para plataforma SIP abierta (Voipswitch). Requerimiento: 1 - Selección de firmware compatible con SIP no propietario. 2 - Creación de "[login to view URL]" para provisionamiento remoto.

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    zoho's phonebridge does not fully support asterisk 13, probably because of java issue causing the plugin not to send the correct information to zoho i need to be able to get full feature set from the phone bridge : caller id + popup + call duration once answered click2call + call duration once answered all call data should be listed in the crm based

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    ...getting a dhadi/Asterisk/POTS configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work properly. Config files and CLI output: [login to view URL] Dialplan: [login to view URL] The machine once finished will backup and replace my old Asterisk servers that basically operate as POTS PBX on my office LAN

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    Install Issabel PBX Đã kết thúc left

    Install Issabel PBX and make it work with my phones

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    Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.

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    Delphi sip client Đã kết thúc left

    I want to get Delphi sip client. delphi version is 7

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    I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer • Blind Transfer • Supervised Transfer •...

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    ...if VoIP package has enable, if enabled then execute real-time billing logic with client package rate and remove credit from credit balance. If client’s package has free minutes, check if over minutes then execute real-time process. Issue remarks in Manage Credit/Client Description field within table as follows: (you may change your variable names to

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    ...backups. I also need to set it up with a Control panel like Plesk or cPanel. It needs to be setup with a hosting server, mail server, SSL for domains, and I will set up an Asterisk server in one VPS which also needs to be secured. I also need help migrating over from my current VPS to the new dedicated server setup. If you think you can do this please

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    Write some software Đã kết thúc left

    I need you to develop some software for me. I would like this software to be developed for Linux using PHP. Asterisk voip and e1 integration for call center operator.

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    Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own ...languages please contact me with a proposal and make sure to write "iluitech" in your message so I know you actually read the requirements and know that you can do it. If you have SIP experience and know PHPAGI it would be a huge plus!

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    Project for Anton F. Đã kết thúc left

    Hello need a pbx setup

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    Python sip registering script Đã kết thúc left

    Hello, i want script to Test sip accounts with Back SIP response codes Example : HOST = '[login to view URL]' SIP_PORT = 5060 LOCAL_IP = '[login to view URL]' PROTOCOL = 'UDP' USER = '509' PASS = '509123' and it will return me with [login to view URL] (200 OK or 301 Moved Permanently OR 401 Unauthorized etc...) +save

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    Voip sip registering script Đã kết thúc left

    Hello, i want script to Test sip accounts with Back SIP response codes [login to view URL] I will provide : Server ip : Port : Tcp/Udb : Username : Password : and it will return me with 200 OK or 301 Moved Permanently OR 401 Unauthorized etc... +save output into text file +Be able to run in multi-thread Job urgent

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    hello, i have this package : [login to view URL] need to...package : [login to view URL] need to install on my server windows then build api requests to manage users and add sip accounts SO i will be able later to use on my custom cms

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    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well

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