VoIP Jobs

- 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

I am looking to build a comprehensive Fax web-app with the following requirements: 1. Email2Fax.(including Email2Fax to multiple fax numbers at once through an email distribution list) 2. Fax2Email 3. Admin interface to manage users settings etc. 4. User interface to send and view received Fax 5. Storage and protocols must be Secured and encrypted for HIPAA compliance. **Please provide a demo for...

We basically need a regular vendor for VOIP Carriers on regular basis.

I need a systems technician to set up a linux server with firewall, email, VoIP and Messaging services. The server will be either a phial server or a virtual server with a recognized service provider

I am having some issues with recording in Jitsi. Need somebody to help me fix it. Thank you

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

I want to setup ViciBox on a cloud server. I am thinking of using Google Cloud Computing, but if there are better alternatives that are more simple and more cost effective, please let me know. Also, I would like to link my domain name to the server where I can login from my domain name with a username and password. Please let me know how much and how soon.

Urgent requirement of an expert to install and configure the ViCi-Dial for my company.

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.

Job Description: Roles and Responsibilities: Developing feature sets for a cutting edge telephony product on Android. Unit-test code for robustness, usability, and general reliability. Work on bug fixing and improving application performance. Continuously discover, evaluate, and implement new technologies to maximise product performance and robustness. Maintaining technical documentation like arc...

I'm looking for a service to install Asterisk or Elastic PABX for my call center Must have recording feature and reporting module, also looking to integrate this PABX to existing in-house CRM

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 hum...

Hello I have issue with my Asterisk certified/13.18-cert2. it 's rejecting the incoming call with the header Tel: ... I need a small modification on the asterisk to accept the calls. I need developer who will develop small patch for me . find the attached invite

Is it possible to create an app that can find out who is originating spam calls? I want to identify the company calling me so I can contact them and let them know I have their identity and want them to stop calling me. With spoofed numbers it's very difficult to talk to anyone in charge.

Need to properly configure the router Linksys WRT3200ACM. The connection of 2.4 works perfectly but the connection of 5 Ghz no Internet Afte need to Turn off the DHCP servers in all other devices

IVR Industries (and sub-industries) Manage audio lists by industry 1. Menus - each menu has playback file to navigate the listener what button to press. 2. Menus - ability to make sub menus. 3-5 levels. 3. Menu can navigate to: a. Sub menu b. Playlist of files located in the server c. Connect to Icecast Live Broadcast url to listen to radio. 4. Ability to insert "PROMO" files before ...

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.

I'm using OpenSIPS with SippySoft. I want someone to update the OpenSIPS to latest without loosing any functionality and show me how it's done. More work will be provided in case of success.

Hi. I need someone to configure STUN & ICE on asterisk. More requirements will be discussed over chat.

Hi, We are in process of building app on Android. When making audio video call users can't hear incoming or outgoing calls. User can't hear other user as well. We are using Twilio Api It was working fine but now app is on play store we are getting this problem. Need urgent fix only who is experienced. No time wasters

I need to configure asterisk to hide or mask the RTP IP of the vendor by masking the IP with a name. Please let me know if you have experience in this.

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

We are looking for contact centers for a Customer Support campaign; The Contact Center shall handle their inbound calls from customers regarding the Inquiry of products, Complaints handling, Wrong product delivered, Refund queries, Register refund request, New / upcoming sales or offers, etc. -Seats Required- 20 minimum -100% Advance Billing given by client For more details please send a messag...

kannel expert don't waste my time , I have two apis one to send sms ( we need to save variable name mms) second we need top dlr-url=%mms that's it I need someone who knows how to parse

I need SMS setup on my FreePBX & FusionPBX hosts, I also need documentation and training on the process.

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 need an Asterisk Expert for couple of works. More details will be described later. You need to be able to create asterisk scripts and have to be expert on making dialplan.

We need someone who can find our call centre new projects to work on, in order to be paid you will need to provide me with a contact for a new project, once the deal has been done and we get our first payment you will be paid.

i need integration erpnext crm with issabel and make pop up on call show customer details and call record and ather and i need sorce code We should have incoming call popup with data being picked up from the ERP Next CRM Module for a existing number and We should have a option to save a un-known new number as a new Lead or Save to a existing customer / supplier / other contact similarly we should...

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 Tena...

Need a tech or developer to tune up Fusion PBX [login to view URL] some registration issues on Legacy devices like Cisco [login to view URL] issues with ATAs, Cisco SPA 112 & Grandstream ht801. Also need to whitelabel fully.

I know there is no OOTB or Free way to rebrand FreePBX. I am looking for a way to rebrand anything at all for FreePBX through code or developing: Logo, icon, names, footer and headers.

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

Hi We have vehicle rental business where customers call us for vehicle booking. We have multiple locations . We want to set up inbound calls center preferebly 24 hrs if possible.

We have a ISR4331 that we use as a voice gateway. Currently it is on version isr4300-universalk9.03.16.04b.S.155-3.S4b-ext.SPA.bin. The TCL script send a tone to an FXO port at certain times that we use as a shift-bell. I would like to upgrade to [login to view URL] but when I do the TCL does not function. Because this is a production system we need to troubleshoot during non-business times....

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

Please let me know if you have experience in above tools. I've some little things to do. First project is. I need to mask the RTP IP of the originating site with any IP which is not present in the server. So the termination site will see a different RTP IP than the original one. More details will be discussed on chat.

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

Le projet est la création d'une plateforme de voyance en consultation privée par téléphone, tchat et mails. Il faudrait que le client puisse acheter des minutes de consultations dans un portefeuille et qu'il puisse les utiliser en téléphone ou en tchat ou les 2. Il faudrait qu'il y a un module d'avis que le client puisse laisser son avi...

Используем сервис RingCentral (телефония по США), на сервисе порядка 100 пользователей, которые работают удалённо (страны СНГ). Столкнулись с рядом проблем, но не можем разобраться - или это на стороне сервера (сервиса RingCentral), или это на стороне юзеров. Нужно изучить статистику по качеству связи, проверить "проблемных" сотрудников на предмет качества подключения и дать отчёт по н...