Sip dialer symbiantrabajos

I need to pull data from emails from 5 different lead sources and send it to a CRM , MY auto dialer and my texting company it will involve coordinating with several people.

I need to pull data from emails from 5 different lead sources and send it to a CRM , MY auto dialer and my texting company it will involve coordinating with several people.

Currently we have a project to interconnect SIP Trunk from telco provider with Yeastar PABX throug Mikrotik. Up until now it still does not work and we need netw. engineer to solve this issue.

...would like some help getting the cloud software configured to connect to my extension at home. The software is is installed and running on a commercial (Crown) cloud environment. I need to: 1. Help me access and set-up PIAF with my one extension in my office. I don't have any port blocking issues to my knowledge for port 5060. 2. I would like to set-up an extension for a couple of cell phones and a SIP adapter for one wireless home phone. 3. I would like some help getting the basic IVR solution running (press 3 for sales etc) There is no big hurry on the project as it is my personal system and I would just like to have it working. Please provide an estimate of time required. I will provide access to the server whichcould be tested with a local softphone installation. **Ple...

A list of US homeowners with US phone numbers and property details will be provided. The script for the live calls and the script for leaving a voice message will also be provided. Instructions on how to highlight the numbers and names are on the script. The dialer for making the calls will be provided. For every thousand homeowners this project is paying a fixed amount between $10-15 depending on the number of appointments received.

I am looking for a Native App developer for IOS and Android who can create a SDK file for our Dialer screen to host it in App of our current product.

Приветствую. Я живу в многоквартирном доме на 150 квартир. У нас вызывные панели BAS-IP с поддержкой SIP. Работает это все через малинку с True-ip, но он очень плохо пропушивает звонки на телефоны (в локалке проблем нет) Немного полистав инет, пришел к выводу что связка Flexisip+Linhome стало бы решением для жителей. У меня есть сервер на площадке ISP. Если Вы опишите параметры, то с предоставлением доступа к виртуалке проблем не будет. Подскажите, на каких условиях мы можем это сделать?

My name is Alex. I live in Kyiv/Ukraine. We have residential building for 250 apartments. Trying to make an intercom system for neighbours with entering panels that support SIP. Searching on the web I've discovered that the Linhome app is most suitable for this purpose. I think that Flexisip server will be correct suite to make it work. Is it possible to ask you to setup flexisip server for me with simply preferences (push enable) and teach me how to add/edit users on it?

Hi if any has previous experience please let me know I want something should work from existing phone number NO hardware required

I need some help with my business. IVR SYSTEMS using freePBX and SIP

i need a sip calling app that based on the linphone customization, and also with the source code, if some one can do it, just contact me.

We need to reconfigure the Cisco system from ISDN2 lines to SIP trunks.

(PLEASE READ THE ENTIRE POST BEFORE APPLYING) Hello all We are looking for call centre with proven experience in SOLAR project in USA RESIDENTIAL. Fresher may also apply, Minimum Number of agents required: 5 Agents to start with The call center need to have his own dialer and VoIP. Home based team can also apply however they need to have their own dialer. Kindly apply with your proposal mentioning 1) How many agents do you have to start with? 2) What is the name of the predictive dialer you are using? STARTING STEPS After contacting us, and we interview and approve each other, afterward training will be provided to your agents. CRM provided to track lead status HAPPY BIDDING!

...computer but I would like to place a SIP proxy and run a webphone in the browser. For that I will need an executable docker-compose file or detailed instructions for: - Frontend - inputs SIP parameters as ip, login and password - button to connect - disconnect button - button to answer the call - button to end the connection - no need to make calls, just receive calls - it doesn't have to look nice, I will adapt the frontend to our internal application. - you can use sipjs, jssip or any other open source SIP library. - backend a backend application to access the frontend, it can be nginx but it needs to be configured - an example asterisk server - running on port 5060 with UDP - a configured extension - SIP proxy / gateway ...

I have SDK file . i need to customize one thing on it Just need to know when i make a call where is that point when the other person answers ? and when he hang up? means need to know where is the point of call duration start and end. Because I'm creating Auto dialer with audio file so i need to know when the call starts to play the audio file. Everything is ready and perfect but this issue .. the audio starts once i click on call. Thanks Thanks

VERY URGENT: NEED IT IN MAXIMUM 6 HOURS PLEASE WHO CAN DO THAT BID Very Simple: Upload phone numbers list. Start Auto voice dialer. When the other one answers the app play a audio file, when finish or he hang up, call the next number Note: i make call throw my PC by connecting my Phone to the PC remotely by this software (SDK Available): Thanks

- Setup a dialler on my server - further setup the dialler's campaign - dialler with recording is preferred (3rd part app with recording is also welcomed) - find a VoIP sip for this dialler's campaign with an unlimited budget plan for both inbound and outbound calling

Our company is seeking a highly motivated, well-spoken individual who has excellent phone skills and the ability to set appointments with ease. Most of this position’s time will be spent on a computer and phone making outbound calls through a dialer/phone system. We are looking for someone who can convert calls into qualified appointments and closed deals for our team. The main skill we are seeking for this position is the ability to change someone’s “No/Maybe” into a “Yes” over the phone. A successful cold caller must be persuasive, personable and someone who loves to talk to people. You will be responsible for influencing individuals to schedule an appointment with a team member. Responsibilities: · Use scripts to provide information a...

I am looking for someone with both excellent spoken English and Mojo dialer experience to generate leads. No selling involved, simply offering info to homeowners.

my website is developed using CMS caller Perfex and I need you to integrate softphone so that my team can call leads using CRM only. Softphone Type: -with SIP login - work with VoIP channel with call recording

Create PC Windows App, C# .NET VB.NET VBA/COM Auto Dialer App Very simple: simple window with Phone numbers file .crv Import function and button to start the auto call and when the other phone answers i need function to play a audio file, and when finish ot hang up cal the next number .. Call gateway is mu personal mobile its connected to my laptop remotely buy desktop and android app (SDK for the mentoring software available) Thanks

Need customisation in android firebase code. In this the data's were saved by json file. Convert html code to json and upload in firebase database. This is just my html code <td><a href="tel:0123456789">0123456789</a></td> "All works fine but one problem in offline mode when click on the phone number the dial pad will not open. " Need click to open phone dialer feature in offline mode also

Hi. I need someone to manipulate the receivers response to the caller according to the time of answering the call. Please let me know only if you know how to manipulate "SIP Response Code" in Asterisk SIP Trunk. Thank you.

We are having some issue with OLD Analogue Phones. The call is answering right after the dial is initiated. So our customers are being billed for hearing the ringing tone. The caller gets the 200-OK message at the beginning of the call. None of above is expected. We need our customers hear fake Ring back, not billed & their CDR not updated as answered until the call is picked up by a real human. That's pretty much it. Please read it carefully and bid if you want. Budget $100. If you have any question then feel free to ask after reading the details I already provided.

...single central A2billing system with a facility of DID management. The trunk which this SIP service is running are directly connected to the A2billing or a gateway attaching a physical SIP/ISDN cables routing back the calls to A2billing. Each Freepbx system has a customized call reporting system that supports call center reporting and call recording with a unique dashboard base system. When a subscriber signs up with an extension, they can access the webservice to see their call center reporting and access web phone. This is available in the system now. The current new plan is to add a new Kamailio system to support the FREEPBX machines >> A2billing route which is FREEPBX machines >> KAMAILIO >>SIP TRUNK. Engineer must possessed the relevant...

I am investigating on setting up 3CX for our company's needs. I am looking for a freelancer to help me set it up. I am a techie, but not an ex...environment does not allow SSH logins. (If I am mistaken and it does, hosting at 3CX is ok). Requirements: - Configure fixed DNS on Amazon EC2 / 3CX instance - Allow 3CX instance to send e-mails (I have Amazon SES account) - Configure 3CX according to the instructions document (see attachment) - Configure 3CX with Ringdesk (details in ) - Configure SIP trunks to allow outbound calls - Make sure 3CX can send e-mails - Finalize by making and receiving a call using softphone on Windows and cell phone I would prefer to work on a shared screen using Zoom. (You give instructions and I'll share my screen)

Hi. I'm running a SIP Trunk with asterisk. The problem is when we call a Analogue Phone the duration starts at the beginning of the ringing. We need someone who could setup the Ring Detection module of asterisk to bridge the call and let the duration start only when the ringing stops. More details will be described with the selected freelancer.

We are seeking for a...for a B2B appointment setting project. The project will require the call center agents to call hospitals based in the United States and schedule appointments for those interested in our company's services. The goal is to get the decision maker's contact details, identify their needs, and schedule them for an appointment with our inhouse sales team. Requirements: Call Center must have their own dialer Agents must have prior sales or appointment setting experience We pay $10 per valid appointment and an additional $15 if the decision maker signs-up (converted by our sales team). Payout every 5th and 20th of the month. We have leads to call but you can also source your own leads as this data are very much available online. If interested please rep...

a way to launch a phone call on with 1 hot key so clik on phone dialer ...click in text box .. . and 1 hot key to clik on end call

Cold calls, Need a predictive dialer. dial 100s of contractors and ask if they do work in our area and then just say "ok ill have the boss call you in 2 min". a lot of the phones dont get answered so we need to try them repeatedly through the day at different hours

I need someone to help me preventing the 200-OK Message passed to the dialer in some cases. More will be described in chat.

I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)

I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)

Need someone who can modify a client software for unified communication client software. Has knowledge of SIP client, video conference compatible with Asterix and Cisco CME

Requirements: - call reject/hang up, - call transfer, - no activity detection/silence detection, - DTMF detection, - Linux platform kernel version from 2.6.

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Nero Drinks Company is a family run business. We produce Nero Premium Vodka, a handcrafted, small batch potato vodka made exclusively of British potatoes. Nero Vodka is velvety in texture and smooth enough to sip neat. Equally its neutral taste and slight natural sweetness makes it ideal to mix in a cocktail or long drink. As a brand we see ourselves as the David to the Goliath of the big premium vodka brands. What we lack in size we more than make up for in determination. Our strap line is the 'Undefeatable Spirit' and our mission is to celebrate the undefeatable nature of our customers and consumers. Our target market is the 25-40 age bracket of males and females who like to celebrate but prefer quality over quantity. We advocate responsible drinking. Our produc...

hi i am looking for a dialer expert who can setup our dialer for us and help run it.

we want to setup Bitrix24 with Sip Connector integrated to our IPBX

Looking for a determined sales specialist/closer for real estate transactions. TRAINING PROVIDED. Responsibilities: Make Calls to Potential Sellers (warm leads) close contracts, make follow-up calls, and schedule closings. Real Estate experience preferred and/or Sales background WORK FROM HOME TRAINING, SCRIPTS, DIALER AND NUMBERS PROVIDED

We are a professional product development company with several projects. One new project is a client with the idea to incorporate a aerator into a wine glass - so when the customer uses the wine glass, it puts air into the wine, giving it a better taste and experience. Simply poor the wine into the glass, and then when you go to take a sip, the wine passes through the decanter or aerator, causing the wine to take on more air - this improves the flavor and texture. We will do this with glass so any cool ideas you have, let me see them. Looking forward to working with you. Global Product Makers

I am looking for an audio , SIP , WEBRTC, EXPERT to build a product like 1000% of features and operations. with some additions. with my own dashboard design and home page layout. You must know sip snd webrtc. This is NOT for first timer. I will NOT pay for you to learn. Apply ONLY if you have done similar and can provide sample link and code. If you don't like to listen and follow instructions DO NOT APPLY. if you are inexperienced and cannot work under pressure DO NOT APPLY.. This is only for serious professionals who are willing to work long term and not try to make money now and disappear. I am in this business 40 years and have heard all. I will provide sample layout of homepage and dashboard to NOT CONTACT ME IF YOU DONT HAVE SAMPLE OF SIMILAR

Hi. I'm looking for someone who has ALREADY BUILT a WebRTC app that I can buy, then pay to have it customized. I will only consider those that can send me a login of one they have built so I can test it. Here are the features/requirements: - Softphone that connects to my SIP - Chat/messages which works via API with my carrier - Video which works via my PBX/WebRTC - File sharing on chat - Screensharing while on call - Mobile apps (iOS and Android)

softphone that will register to my sip PBX as an extension. I'll load the EXCEL sheet with leads that I want to call to. the dialer will call them one by one. when it gets sip event of 200K (so it's a valid number)- I can set in the settings if I want to receive the call on the softphone- or it will transfer the call to another EXT or ring group that pre related in the settings. click 2 call big advantage activated by code! (i'll get 1M random codes in EXCEL)- 14 days trial.

Hi, I need support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installe...support setting up a basic on premise (RasPBX) Asteriks installation. I have the following devices: - Raspberry PI 4 with RasPBX installed - 2N technolgies Doorbell with Sip Support (works already and is registered to sipgate) - Several Devices with Sip Softphone installed - Home Assistant on several devices (I want to connect via webRTC (Browser) to the Asteriks) So I need: - basic configuration of Asteriks / FreePBX - 2-3 extensions / endpoints for the devices / softphones - 1 extension for the webRTC connection (e.g. ) - 1 configuration for the 2N Doorbell with SIP Support

We are currently facing VoIP issue on Mikrotik RB750G3. The incoming & outgoing calls can be done but no voice from both sides come out. The PBX system is NEC SL2100 & the SIP Phones are Flying Voice IP622CP.

need a cloud pbx that can be connected to my grandstream ip phone. I need to be able to get calls in one area and then calls can be rerouted via extensions to other phones. This will include softphone solution that can act as a call centre. Multiple lines must be housed. This will not include SIP. It should be able to run all on the cloud and be housed on our server.

I need a cloud pbx that can be connected to my grandstream ip phone. I need to be able to get calls in one area and then calls can be rerouted via extensions to other phones. This will include softphone solution that can act as a call centre. Multiple lines must be housed. This will not include SIP. It should be able to run all on the cloud and be housed on our server.

looking for customizing Linphone for iPhone and android, Need "AUTO LOGIN WITHOUT USERNAME AND PASSWORD" (username and password from a2billing table), PUSH NOTIFICATION SHOULD WORK TO RECEIVE INCOMING CALLS AND FCM MESSAGES, SMS OPTION SHOULD WORK WITH ASTERISK SIP SOME OPTIONS OF LINPHONE to simplefy linphone interface. we have a sample source code and customized linphone dialer TO SPEED-UP THE PROJECT. After finish the project, we need the final source code.

Hi Erewin, I noticed your profile and would like to see if you are interested in helping with my project. I require FreeSwitch / Asterisk expertise to help debug a SIP trunk issue. There is further work with push notification later. We can discuss any details over chat.