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
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!
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...
I have 2 PBXs. The first is not an Asterisk, the second yes. The first use the second to place call with predictive dialing. I want to implement AMD service on PBX2 and automatically drop call answered by machine.
Hello Our current development website is https://didicar.ca. We need to add Civic plugin to the website. [login to view URL] We tried to install several times, but cannot install and found some php errors. If you have experience with civic plugin, that will be plus. If this project going well, there will be more
I need to connect CRM Vtiger with Elastix or Asterisk to make calls clicking on the CRM
Design a logo "PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a..."PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a glass of wine to each student, so we call it paint and sip. We are looking for a fun logo.
I need to connect CRM Vtiger with Elastix or Asterisk to make calls clicking on the CRM
...-Search CDR - with the phone number i can see a record when its called and how many time its called by the dialer. -Download Recordings - I can download here records from the server. -Recordings - i can listen here the whole conversation and even edit the resultcode of this client (what i mean with edit is example: From sale i can set it to Negative or
To duplicate an existing Asterisk Server with WebRTC ( HTTP5 - SSL - etc.) onto another machine and make some changes. Works together with an Apache Web Server and a MySQL Data Base, on separate machines. Must function as an immediate - identical backup !
zoho's phonebridge plugin supports up to asterisk 1.4 with asterisk 1.8 partially working (only incoming calls) with modern asterisk versions (11 and up) not working at all do to (apparently ) java problem zoho's support says that those values : dialed channel, linked channel, dialing channel are NULL instead of being filled with information. we
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"
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
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.
I have a working FreePBX server. The freepbx server is running good so far. The following changes needed in my server. 1. I want to install a open source predictive Dialer in my freepbx. I have have chosen VICIDIAL for that. You may suggest better one. The dialer must use existing extensions for auto dialing features. 2. You must configure the system
...[s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [login to view URL]: Caller ID name is '+918820094576' number is '8902050098' This should be a ver...
...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]
Hi, We are searching for skilled Asterisk developers.
...the moment. 3. Peering to another VOIP server 4. Bring our old server up to scratch. 5. A few more minor ones that I can't remember right now As the aastra phones are currently using the aastra XML login scripts experience on these handsets will be ideal. However will consider ditching the XML and use server-side endpoint managers if it can achieve
...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
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.
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
...We 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;
I bought 1 landline (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
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
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.
...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
...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
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.
I am having problems 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
Существующая сеть передачи данных и телефонии по SIP. Необходимо осуществлять мониторинг сети, программировать коммутаторы, медиашлюзы, поддерживать SBC и т.п.
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 •...
...Prevost, Quebec) Tax Rule rate = 14.98% 2min outbound call on SIP Canuk 200 plan 0.020 = 6sec increment 120sec 2min * 0 .020 * 1.1498 = 0.045992 (round up to 0.045) 3min inbound call on SIP Canuk 200 plan 0.025 = 6sec increment 180sec 3min * 0 .025 * 1.1498 = 0.086235 (round up to 0.086) SIP Canuk 200 Package Detail [login to view URL] & [login to view URL] (already e...
...for a server and security specialist who can help me set up a dedicated server. I need to create several VPS server on it and set everything up with security and 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...
Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own extensions and modify their call routing. I need a developer that is an expert in Node.js as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with
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
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
hello, i have this 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
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
VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist
hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [login to view URL] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [login to view URL] will be long term project if satisfied with the [login to view URL] budget is 100$ for this project