Very Easy Lead Generation. Work From Home. Data, Dialer, Voip will be provided from our Side You need to talk to resident in USA. Hourly Wages
we need a Predictive Dialler with freepbx to link to Qs and have a wall-board with reports. reports need to link to recordings. this all has to work on freepbx and sip and softphones. The PBX will make the call and the agent when signed into the Q will receive a call. the agent phone rings first then the PBX makes a call according to a predefined list
We are building a state of the art android phone dialer app that adds a call assistant to the incoming call experience. The app is a full call take over (both incoming and outgoing). We in the process of doing a private beta test and getting lots of bug fix request and want to add new features to enhance the user experience.
Hi I need an application that can receive a call through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset and set it as default
...to (IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2 5-
...verification in VoIP softphone and php backend, create IOS and Android softphone General project description: We are running an A2Billing + Asterisk server and want to have a branded Softphone for Android and IOS. Registration and provisioning of softphones must happen with SMS verification. For the softphone we want to use the source code from VoipGrid
...1main server other all asterisk server work like VM host1 IP [đăng nhập để xem URL] SIP 5060 DB NAME datastore1 host2 IP [đăng nhập để xem URL] SIP 5060 DB NAME datastore2 host3 IP [đăng nhập để xem URL] SIP 5060 DB NAME datastore3 host4 IP [đăng nhập để xem URL] SIP 5070 DB NAME datastore4 host5 IP [đăng nhập để xem URL] SIP 6060 DB NAME datastore5 i hope y...
Hello All, We need to integrate Vonage and Callcentric SIP with our Bitrix24 installation for calling to USA. This will be used with a 10 PC user system for a call center. Other requirements with same: - SIP set up in Bitrix24 - Setting user right for 10 users with extendable option to more users. - Setting up rules for call forwarding, call picking
We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Proficient in Lantek program (lazer tube cutting) is a must. We have ongoing work for the right programmer on a long term basis.
...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Artificial Intelligence: It involves Studying Human thought processes representing thought process on machines. Knowledge encompasses the implicit and explicit restriction
...competent programmer who is able to build machining routines for our 5 axis router. We are looking for someone who preferably has experience in processing large timber panels or Sip panels. Our models are prepared in Solidworks and require converting using HSMWorks software. We are looking for someone who can optimize our cutting routines and prepare new
...Installed. We need an Expert in VICI Dial who Can 1)Install fully functional Vici Dialer on dedicated [đăng nhập để xem URL] Clusters on expandable Seats with Database, Webservers. 2)Configure Softphones with Dialers and Train Call Center Agents. Create Agent Logins and Vici Dialer Admin and Create Campaigns,Upload Leads into the Hopper etc etc. Applicant Must
I need help completing a few tasks with my 3cx system: 1- i need to update the default template for a yealink t46 to accomplish the following: a- I want to set the BLFs based on a static list of extensions (i.e. 100-110) which can be hard coded into the template but i want the template to read the associated names from the extension list and label the button with the [hard coded extension]- [e...
...When number clicked on "Free Call" button, user get connecting message and generate 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...
...tecnología VoIP y siempre esté conectado con el resto del mundo. Con mi experiencia puede disfrutar de los siguientes beneficios: -Administracion de 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
...budget is $30. Lower bid will be awarded. I'll like to develop a secret caller id app, so that the called person do not know the caller number. There are 2 tabs on the app, dialer tab (to call people), and call history so all the calls within the app are stored there, it's possible to recall from call history or delete calls. There are google ads on
We are looking for candidates who have prior experience of USA dialing. This is Work from Home opportunity. Data, Dialer, and Voip and Process Training will be provided. This is very simple lead generation process for Energy. We are open to paying on login hours to deserving Candidates, after testing them for a week. Very High Earning Potential..
I NEED TO IN INSTALL A NETWORK VOIP SWITCH WHO WORK IN BOTH DID AND VOIP ALSO CALL RECORDING FACILITY CURRENTLY I AM USING ASTPP BUT THEY CANT RECORD ANY CALL
Hi Olexander 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
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
HI We need to have a SIP trunk between a Telegram ID ( [đăng nhập để xem URL] ) and a SIP server, so every body who calls our telegram ID using Telegram messenger, will be forwarder to our SIP trunk,
We are facing some challenges regarding our "On Guard" phone services - we need a solution which can provide 24/7 Phone support, Insights and a really good customer / user experience. 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...
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
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.
Looking for someone who are expert in fixing the asterisk server and...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@[đăng nhập để xem URL]>' failed for '[đăng nhập để xem URL]' - Wrong password " is happening.....
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
I need a module which can initiate a direct call (Chat/Audio/Video) to a given phone number via imo App. Just imo contact list numbers will be used. The source code (with detail comments) is needed and could be run on android studio. [đăng nhập để xem URL]
Hello, I need an application like this: http://ip/[đăng nhập để xem URL] 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 id spoofing. Very simple. I will give you server information. Thank you.
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...
Setup, configure and test voip server Server setup and hosted on MS Azure Basic 3CX PBX setup already. Need help specifically with the following: Firewall configuring Provisioning (yealink t46) UDP registration Conferencing setup BLFs (presence not working, assuming it’s the firewall/ UDP) Formatting of display (caller ID formats etc) Shared
...interested on making good money feel free and apply Need to have experience with a good voice articulation This role the priority is help the client from the beginning to end. someone who will do follow up to client. Organized person. Customer services Driven. Regular agent are doing 4 closing DIALER+DATA+VOIP PROVIDED C.R.M. PROVIDED
...addition to or instead of DTMF tones by way of the FreeSWITCH mod_pocketsphinx. https://freeswitch.org/confluence/display/FREESWITCH/Mod+pocketsphinx [đăng nhập để xem URL] Eventually another ASR will be used, but FreeSWITCH has a uniform API for interacting with ASR modules. Linux Debian 9 development virtual server with environment will be