we need the sms and IVR developer for our projects the below details are here: SMS- suppor...send and receive multi SMPP account billing for each transaction log generation filter and rules apply DND (Blocklist Feature) connected with MySql IVR- IVR flow CDR sip account call record billing option the criteria based on portal or application
We're looking for someone who has previously written modules for Asterisk. Modules that preferably were constructed to work in version 11 and 13. The ideal candidate will be able to stand up their own Asterisk development environment to build and test in. An asterisk module which will allow us to, while placing an outbound call and playing a set
...a VOIP project which requires SIP Trunking methodology and integrating SIP Trunk with Asterisk server. (Asterisk version is important, while you posting at which version can you able to deploy) - Need to have excellent skill set in Asterisk, PBX, Integration of SIP Trunking - I must able to make calls thru Asterisk. I will give you more...
Need a 3D rendering of a Ford f450 pickup truck, with custom trunk that includes a hanger and a trailer. (Please refer to the images attached.) This 3d rendering is fairly short, about 5-8 seconds that shows the entire model in a 360 degree view rotation with a transparent background. We need this done really quick, before the end of the week. Thank
need help fixing a vicidial server when the agent logs in the initial call doesn't come over for 45-60 seconds well after the interface times out
...design that is similar. All the pricing and details can be found on the website: [login to view URL] On the flyer I need date, time, location, ticket pricing (with an asterisk that the prices are early bird pricing) I also need room to add Sponsor Logos which I will add later. The website must be listed as well. The tickets should have all that
... Profile picture upload issue 2. First Name and Last Name in 2 separate boxes should be instead of full name. 3. Fields validation issue (marked by red asterisk) 4. Target Job location (the list should include drop-down menu with Qatar cities and zones, multiple choices option should be available) 5. Job Industry, Career
...Is Coding Asterisk Virtual Server. 1: Code Asterisk Server And Configure To Be Used In A Local Area Network With A. Hard Phones B. Soft phones Application Installed In Android Phones IPhone Phones N:B The Coder Will Recommend The Hardwares A. Asterisk Hardware Server
...telephony integrations. Proficiency in VoiceXML programming, Core Java/ C#.NET, Avaya Aura Orchestration Designer, Avaya Aura Experience Portal, Lumenvox Provides strong knowledge of Call Center Technologies (Nuance, Genesys, Virtual IVR…) including CTI applications using Genesys, Avaya PBX and MS SQL. Maintain and manage Genesys Routing, Framework and reporting
Need install asterisk and sip server on virtualbox for receive calls directly to computer and if 1st line is busy automatically forward to second free line with call recording and without monthly fee
I have a production asterisk installation running on my server. I have a requirement. I want to setup a queue such that Agents and end users can use queue using their mobile phones. Lets Say, their are 3 agents Agent 1: Mobile : +91-XXXXXXXXX1 Agent 2: Mobile : +91-XXXXXXXXX2 Agent 3: Mobile : +91-XXXXXXXXX3 Lets there are 5 users who will dial
We are using the Asterisk PBX With a Linphone SIP client in a Linux environment operating on the Olimex A20 and PINE64 and are experiencing very high echo. We understand Linphone uses the Speex Echo Canceller. We either do not know how to adjust the echo canceller or we need to substitute it for another excellent echo canceller. We would like someone
need a Technical with experience with IP OFFICE CONTACT CENTER 10 to Implementation the chat server and what is the requirements.
I am looking for someone who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the re...who can customise the UI on Linphone for iOS, Android, Windows and Mac in that order. We will be providing all of the required graphics and we also have an internet facing SIP server it will register to.
...(inclusive) and store these into an array. Produce a chart EXACTLY like the one below that indicates how many values fell in the range 1 to 10, 11 to 20, and so on. Print one asterisk for each value entered. Notice the spacing for everything. Range # Found Chart --------- ---------- -------------------------------------------
Hi altr, I'm looking for someone to help me r...using freeswitch for wholesale scenario, When i receive 181 call is being forward sip message from supplier, it is being absorbed by freeswitch and not sent to Leg A. I don't want to spend time to figure it out, i just need you to tell me how i can make sure this SIP message is forwarded to the customer
For this project i need some assets to be textured, I provide you t...assets to be textured, I provide you the model with uv unwrapped, you just have to texture it, Mostly military wood crate texturing, i would need 2 variant and wood log/bark/trunk and plank texturing. Best workflow would be pbr for Arnold 5 and using displacement and normal map.
...be answering several questions and my answers will be focusing on 3 issues. I want the ppt slides to be mainly around a tree with three clustered branches/leaves; a defined trunk; and lots of roots. I want the slides to only contain key words and pictures with max 5 slides -- (1) Full picture of the tree; (2) Picture of one set of branches/leaves labelled
We're looking for a senior React Native/Redux developer to complete the final steps in a VoIP/Text Messaging mobile app. The mobile app uses PJSip to communicate with Asterisk, and interacts with a back end API created in PHP. Ideal candidate would have knowledge of React Native, Redux, and PJSip (Optional but definitely recommended). The final stages
...script will be provided but you will have to update it with notes. - Qualifications: I need someone that speaks great English, has organization skills, experience with VOIP (SIP) or has a similar program already so that we can check. Duties: Negotiations with the top officials of the companies; Sale of services in the field of B2B; Requirements:
Hi , I am looking for someone who is able to configure ZRTP on freepbx ? and would like to know if all features work with this protocol? I read some articles said that call recording is not possible with ZRTP. Does it work with all other protocols such as, TLS, UDP and TCP? Let me know if you are interested in this kind of work and let`s discuss the duration and price.
We want a site where we easily can see the calls that come in and what happens to them. An example could be A customer calls 70209404 (NordicCall), the customer is in the queue for 2 minutes because every agent apart from two are on DND, one agent is busy an the other agent rejects the call. So we must continuously be able to see all of the information, and there is a site with further informati...
...and, if they are valid in system. If so the backend server would send call set up info (dial plan) to asterisk server (I have an asterisk Guy to work with). The app would then dial an 800 number that was returned to the app on call setup. Asterisk would match the customers CID and process the call as per the ad hock dial plan created for that call substituting
Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat Livecalls IN/OUT statistics Calls report cdr Audio file Create Survey Phone book where can upload
...with in my budget don't waste your time on chat if you bid less and then increase it after dont bid Hi Dear, looking for developer who have experience related to voip asterisk freeswitch php for develope dialer for incoming and outbound calls for run voice campaigns there will be few features in dialer which i can explain more in details via chat
Hello, We need to develop a SIP to Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows/Android, or by using the
Hello. I need to implement a click to call system on my website. I've a list of tech su...can be billed 10€ for 10 minutes or 20€ for 20 minutes. Only after the site deductes the credits, the call will start. It needs to connect to some cheap voip server (3cx, asterisk, freepbx, etc). The support tech person cannot see the client phone number.
Install VPN and setup a server connection Install OpenSIPS with graphical user interface install codecs connect GSM gat...connection Install OpenSIPS with graphical user interface install codecs connect GSM gateways to server setup OpenSIPS billing module install fail2ban and configure configure sip connections with clients and perform test calls
...Admin - login access Agent - login access Customer - login access Admin - Features Add, edit, delete Customer accounts Assign customer phone numbers (integrated with Asterisk ami to enable screenpop) Add, edit, delete Employees (no login) Add Customer business info (screenpop, location info) Add screenpop Forms Add changes reason text box.
...end provider (SIP or otherwise) and server details.(ie. Centos with WHM and cpanel running Asterisk) I already have some hosting options in mind and I prefer Centos with WHM and cpanel, running various services to accomodate the VOIP server and the website. Basically, we need to figure out what voip server and back end sip or trunk providers are available
Skype Connect has the SIP trunk feature to use Skype as a SIP trunk of PBX. I tried to configure Skype connect with FreePBX but couldn`t make it work. If someone can do that and already configured such setup, I`m ready to pay.
...those shown in "Sedan [login to view URL]". Please also remove the center post between the open doors for the picture. In addition, please add on the old school trunk. This is shown in picture "Rear Trunk Add On" Third Picture: This picture will be as if you are sitting in the back seat looking up through the sun roof at the New York High Rises. I have
...from iso on a bluehost server. I can register my sip phone, I can make inbound and outbound calls. The only problem is there is one-way audio on the call. If I call from my Windows Sip phone to my cell phone -- My cell can hear me, but I cannot hear the cell. If the cell phone calls my Windows Sip phone -- The cell can hear me, but I cannot hear
One Server with multiple disks managed through KVM - qcow2. All disk OS are Ubuntu 18.04. Security is important. To check the server, it takes ...To check the server, it takes a late Teamviewer. Requirements: Nginx rev. Proxy, Apache2, Certbot - SSL, SMTP server, HTTPS, PHP, MySQL, KVM - qcow2, LibreOffice, Ubuntu, Asterisk PBX for invoice info., etc.
I am looking for provider who provide me sip gateway for Indian operator but with open caller id and API. My concern is to send broadcast pre recorded voice.