Voip app blackberry asterisk jobs
Selling voip connectivity and pbx solutions to residential & business prospects in the UK. Remuneration will include basic pay per hour + great commission
Hi Eremin P., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
Hi Vinod Kumar I., I noticed your profile and would like to offer you my project. We can discuss any details over chat.
We are a Voip based company we are needing someone who can search and find us partners to add onto our routes the job entails finding providers and signing up with them so we can use there test account service
Looking for a call centre who has experience in the US B2B Energy market, pay per sale model which can move to an hourly rate provide a trial period is successful. Please only apply if your a call centre and have all your own infrastructure i.e dialler, voip etc
ekh custom software benana he jisme ki me mera 200 students ko ekh saat call keru per call me sirf me use software ke aunder ekh voice bolunga jo record hoga fer usse use software me daalunga aur fer 200 students ke mobile number use software me daalunga aur send kerunga tho aur time set kerunga aur send kerunga aur sebhi ke pass ekh recording call jayega jisme mobile number bhi automatic dusra hoga ye ekh number he ish per call ker ke demo dekh lo bilkul esha call hota he custom software benana he +918045686530
hi , hope all is well. i am looking for someone to help diagnose the connection between my asterisk and sip provider
Need help building PBXact and custom Modules. Should know how to review PBX Status on Asterisk Server and more. We are looking for somone with past experience with Asterisk.
Hello, I am looking for VOIP expert who can assist with creating an auto dialer campaign via 3CX V16 with enterprise license. The need is to call defined calls list and assign calls to specific calls group and assure calls are out via predefined caller ID
Hi Ankit G., I have a VoIP app developed in swift based on linphone sdk, I am facing issue with callkit implementation and need help urgently. Please let me know if you can help me with this. Thanks vidhya sagar dixit
Hi, I have an Iphone VoIP app (written in swift) used for making and receiving calls. I am facing issue with it. when callkit ui is presented on incoming call and you answer the call it does not get answered, instead the call gets connected for person who is dialing but the UI stays for the user who is receiving call. I am using linphone sdk for voip. Please bid if you can fix it.
we need configuration of freepbx virtual machine/asterisk we have about 3 phones to configure there
I want to extract contacts from google and outlook to skype and ms teams Need someone with experience In your cl write blue so i know you read
I need to create an intro video for my courses at VoIP School The tutorials are in the youtube and voip.shool. I prefer something minimalist, discrete. In one word the intro should look professional! The video should contain the url https://www.voip.school.
I´m looking for someone who can install and configure sipewise (). Someone who have enough experience about VOIP, SIP, PBX. You sould be dedicated, serious and working transparent and clean. The sipewise will have to be configured for the moment with one phone provider with one "big" SIP Trunk and multiple channels. In the moment some 30 phone numbers. On sipewise I will generate different SIP Trunk based on the request of the client with one phone number and 2 or 3 or 8 channels. This SIP Trunk "generated" and managed by sipewise will be connected in 90% to an other PBX which the client have or in our data center or in his infrastructure. To secure the SIP Trunk we should think how will be the best solution or to offer to the client based on his possibili...
I am having a hard time integrating asterisk with Odoo. I have set up asterisk in the same server as Odoo, also have installed certificates for webrtc as well as for the odoo portal. I will provide access to my sandbox server.
requirement is to make the function to register call directory. Our App is some kinds of VOIP app like TextNow. Our users can received the call/make the call. Also there are several customizers for each user. If the customer makes the call to the user, the user can received the call. But the callerid says "undefined" Fortunately we are able to create our own directory to provide the match. When a phone receives an incoming call, the system first consults the user’s contacts to find a matching phone number. If no match is found, the system then consults your app’s Call Directory extension to find a matching entry to identify the phone number. This is useful for applications that maintain a contact list for a user that’s separate from the syst...
Objective: Just need Skype or or MAYBE ms Teams to sync with different devices. The DREAM is they actually LIVE SYNC with_outlook contacts HAHA! Problem: Ok Skype appears to have disabled contact synchronization across devices since Aug 202. Webskype has NONE of my 4,000 contacts. Can you help me figure this out. Could be happy with if I can get my contacts in there. I see MS teams is FINALLY offering call outs. Doublet if any of these are yet live sync with MS People( outlook contacts) would be happier syncing with Google contacts. Solution: if you are a Skype Expert and have solved some of these sync problems I WILL HIRE YOU!
Small multi platform script that: - Registers with a SIP server - Initiates a call - Plays MP3 files - Does DTMF - Ends Call Must work on Windows and Linux "out of the box". All components and source files must be included.
Looking for Call Generate system for Looping , System Generate calls and it go to supplier and from supplier that call come back to our server if same call come back to our system then only call hold me required duration else disconnect immediately also manipulate ASR and ACD as per our requirement
some DID Mexico Voip & SMS PBX algunos DID Mexico Voip & SMS PBX
need some cloud voip serveurs for swithing calls(fusionpbx,freeswith,elastix)
requirement is to make the function to register call directory. Our App is some kinds of VOIP app like TextNow. Our users can received the call/make the call. Also there are several customizers for each user. If the customer makes the call to the user, the user can received the call. But the callerid says "undefined" Fortunately we are able to create our own directory to provide the match. When a phone receives an incoming call, the system first consults the user’s contacts to find a matching phone number. If no match is found, the system then consults your app’s Call Directory extension to find a matching entry to identify the phone number. This is useful for applications that maintain a contact list for a user that’s separate from the syst...
We are looking to create a centralized Dashboard that will help provide better oversight of our phone system. We require help with creating a phone reporting tool to display call statistics by department for the company. Our current VOIP systems consist of: a Zultys MX-E (Call Reporting Database stored in mySQL database) and Microsoft Teams Auto Attendants and Phone Queues in Office 365. We are new to Microsoft Teams Voice, and would like to create a combined reporting dashboard that will oversee all our phone systems. This project would involve use of Microsoft PowerBI, SQL, Zultys MXReports, mySQL, and other recommended programs. We would also like help with interpretation of desired reports and ability to create the reports that meet those needs. Desired Timeline: October ...
We have a web app based on Laravel 4 and would like to update it to version 8. Maybe some bug fixing is required too. There are several custom controller, views and models which has been individually developed. It would also be good if you are familiar with Asterisk / VOIP technologies, which is part of the App.
We want to records all Telegram VOIP call directly from the app. (and save them or send them) (open source project: You can use this github project if it's helping: )
Situation We are aiming to dial into a Webex meeting from a SIP device. We have a self-hosted Asterisk server that is connected to our video conference application We are registering this Asterisk SIP account with our video conference application and Dialing into Webex cloud meeting and various telepresence hardware such as CISCO Telepresence, Polycom etc. Complication 1. We are able to successfully connect to telepresence hardware with no problem whatsoever. - PASS 2. We are NOT able to connect to webex cloud meetings (both personal account and organization account). ​We are getting we are getting this 408 request timeout error. - FAIL Screenshots attached for reference Requirement 1. We require you to work with us and help troubleshoot this issue.
We are a USA energy broker offering commercial customers cheaper energy pay is between 30-45 days after contract is signed and very high commission. there will be no hourly rate to start with until you can prove you can sell. minimum 10 agents, must have own VOIP, Dialler etc websites can be provided for free data.
...Raspberry Pi and have installed (initially) the Asterisk + FreePBX Per this documentation: It got to the point where I was installing the security packages from the command line and it asked if I wanted to overwrite a certain Python file and realized that if I do and it breaks that I need to start over so it'd just be much quicker to work with YOU! So that I can see how to do this properly. Shouldn't be long and we can actually just do it from the web browser (GUI) - in fact we'll have to in order to screen share. I need the RasPBX - Asterisk Dialer to: - Auto-Dialer - Power Dialer - Voicemail Drop (doesn't ring their line, just leaves a voicemail) All of this is actually already baked into Asterisk+FreePBX so it's really just a matter o...
So requirement is to make the function to register call directory. Our App is some kinds of VOIP app like TextNow. Our users can received the call/make the call. Also there are several customizers for each user. If the customer makes the call to the user, the user can received the call. But the callerid says "undefined" Fortunately we are able to create our own directory to provide the match. When a phone receives an incoming call, the system first consults the user’s contacts to find a matching phone number. If no match is found, the system then consults your app’s Call Directory extension to find a matching entry to identify the phone number. This is useful for applications that maintain a contact list for a user that’s separate from the s...
...Experiencia BÁSICA en desarrollo en Java SE; de no ser experto nosotros podemos asesorarlo. - Experiencia MEDIA o AVANZADA en desarrollo en C++, tanto para Windows 10 como Linux... + Programación orientada a objetos. + Interfaces gráficas UI. + Redes: sockets, protocolos (HTTP, SSL, SMTP, etc.), cliente/servidor, etc. + Multimedia: manejo audio/video, VoIP, drivers, etc. - De preferencia experiencia con TRANSPILADORES a C++. - Que su codificación esté comentada y sus DESARROLLOS DOCUMENTADOS. - Que esté dispuesto a brindar CONSULTORÍA SOBRE DUDAS con sus códigos; pagada, claro. Se adjunta en PDF documentación completa. De existir interés en trab...
Need to build a web base CRM and VOIP Contact Center in react. we are looking for a VoxImplant expert. Type MARIO to verify you are not a bot please.
I need someone who knows about asterisk PBX for packet related issues on Asterisk
Hi Folks, We have a standalone asterisk server recently installed on a Ubuntu box. We want to achieve the following in terms of network connection 1. Ethernet interface - This connects to telecom provider for asterisk line and should be used for everything related to SIP 2. Wireless - We want to use this for connecting to the internet Need help from a networking/asterisk developer who can help us split the traffic in the above-mentioned fashion.
Want to able to transfer a call to another agent/queue using a prefix so we can record a note that is then played to the new agent when they pick up. FreePBX 15 Asterisk 16.13 PHP 5.6.40
Hi, I want VOIP calling system from develop from scratch i mean with all configuration/setup where i can trigger call by using customize caller ID in India.
Thanks for reading, we have a problem to solve as below: 1. We are running Asterisk IP PBX soiftware on MT7628 Cpu running OpenWRT 2. For audio we are using PulseAudio and we have some problems: 2a - The echo cancel algorithm for the on-board speakerphone gives echo to the distant end, i.e. the echo cancel is not fantastic and needs looking at 2b - We are struggling to play recorded audio over our speaker 2c - The PulseAudio always requires a second reboot to work properly. If you are an engineer with solid experience in this area we would appreciate assistance on solving these items.
We are looking for an individual that can operate our shipping department. Their role would be to compare shipping rates, print labels, open cases with the carrier, file insurance claims, prepare customs paperwork for international packages, negotiate rate with carriers, and any other task that is associated with the shipping...insurance claims, prepare customs paperwork for international packages, negotiate rate with carriers, and any other task that is associated with the shipping department. - Prior experience in the USA is preferred. You would be provided with training with our software. We are looking for a Full-time employee that can work from 9 AM to 5 PM CST Time zone. Must be able to call carriers using our VOIP number. Please let me know which Software's you have ex...
Looking for a developer who can make an android and iOS application. Requirements :- 1) VoIP Based 2) User Profiles 3) Payment Gateway Looking for genuine Developers. Don’t ask for Advance as server cost would be borne by us. Thanks, Shivalik Sharma 91-8080252321
Hello: We operate a VoIP telecom platform and are using it with cellular phones as well. We have recently connected a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r48: a...
Hello, I am running a Small call center with Freepbx 14, needs to develop a customer satisfaction survey application for incoming queue calls. Call landed to the queue > agent answer > agent completed the call > should go to satisfaction survey automatically or agent can transfer the caller to satisfaction survey. Custom IVR should be able to play to the caller, should have a dashboard to view the reports. Reports needs to be contain following Date and Time / Number / Agent's Extension / Score needs to be able to run reports monthly / weekly / daily and custom date range.
...doing some operational tasks in the business - like helping with finances sometimes - helping with cleaning my e-mail day to day - structuring my e-mail filters - keeping me up to date about important things in my mail - being my antenna basically for what comes into my mailbox - scheduling my appointments - redirecting e-mails to the right mailboxes - taking up the phone so now and then (VOIP) And basically all that comes with the job as secretary / virtual assistant in order to take some things off my plate, so I can focus on growing the business. Apart from that, we are a young, innovative company. In other words, there is a lot of room for you to collaborate with our team on improving our workflows and processes. We like to give you room for your ideas! We belie...
Hi Josh, I'm a VoIP communications engineer and developer that can add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint as you want.I have 3 locations listed in my file. 2 are meant to load balance eachother. At the moment it works perfectly as the logs show the calls doing a round robin between the 2 locations. We use 2 sip providers. 1 gives us 2 trunks to load balance and the 2nd we use just for failover in case the first provider goes down or is unable to route a call for any reason. Here is my file: The config file I am using was created by someone else for a different system we use to use. # $Id: ,v 1.1 2004/08/10 16:51:36 dcm Exp $ # sample config file for dispatcher module debug=2 # debug level (cmd line:
Hi Josh V., Hi Josh,.... I am an expert VoIP developer. I will add a new SIP Trunk and update existing SMS API to a new API RESTFul endpoint.I have solution for your problem. public virtual HttpResponseMessage Get([FromUri]TwilioRequest request) { TwilioResponse tr = new TwilioResponse(); if ( == "Hello World") ("Hello back!"); else ("Text 'Hello World' for a friendly message."); return (, ); } SMS API enables you to send and receive text messages to and from users worldwide, using our REST APIs. Programmatically send and receive high volumes of SMS globally. Send SMS with low latency and high delivery rates. Receive SMS using local numbers. Scale your applications with familiar web technologies. Pay only for
Hi, can someone do the following: * Remove Logo top left of each page and replace with logo on the business card that blue loop with business name - Resource Appraisals * Change the top of the color of the bar to match...Replace fax line (800) 621-7070 with (866) 906-9326 * Edit the copyright information at the bottom of each page from 2010 to 2002 * Remove the line after All rights reserved with Resource Appraisals and the logo are trademarks of Resource Tech Appraisers Then make the document into a fillable form for Mac / PC, Internet. It will be sent to people to fill it all out and submit it to send back. Red Asterisk are required fields - All fields should support Alphanumeric characters. Under Additional equipment and Notes should support max characters for a long message...
Hello: We operate a VoIP telecom platform and are using it with cellular phones as well. We have recently connected a "feature type" phone using its native dialer to our VoIP (SIP to SIP) Platform (rather than an app on a smartphone). The phone is Android OS and does have a SIP client. The phone has a messenger app for sms and the manufacturer has noted we could make adjustments to the apk as this is publicly available for developers. On our VoIP side the custom application should support the SIP SIMPLE protocol for messaging. It should send outbound SIP MESSAGE to our SIP node IP as well as receive inbound SIP MESSAGE from the SIP node. This maybe helpful … https://cs.android.com/android/platform/superproject/+/android-8.1.0_r...
Hafritech is a technology-oriented company that provides a variety of solutions to its clients through its presence in the Horn of Africa region. The company's solutions include Cyber Security, Network Infrastructure, Cloud Computing, VoIP, ERP, Software Development, and Enterprise Data Center Infrastructure. As a startup, the company is seeking a branding solution for a better professional image.
We want to records all Telegram VOIP call directly from the app. (and save them or send them) (open source project: You can use this github project if it's helping: )
