Look at: [login to view URL] People cannot send me e mail from this form. I want them to send me message to [Removed by Freelancer.com Admin for offsiting - please see Section 13 of our Terms and Conditions]
Fix the [login to view URL] On my site and allow traffic from search engine
...uploading the customer details. Creating and launching campaigns and getting its reports. 3. I have bought an OpenVOX GSM gateway. The GSM gateway is working good so far. But I need a small changes in configuration for smooth operation of the gateway. The gateway is not passing correct caller id to my PBX. Caller ID has two parts. One is Caller ID name
I have bought an OpenVOX GSM gateway. The GSM gateway is working good so far. But I need a small changes in configuration for smooth operation of the gateway. The gateway is not passing correct caller id to my PBX. Caller ID has two parts. One is Caller ID name and another is Caller ID number. Caller ID name part is okay and gateway sends is as the
...retire our incoming ISDN lines and are setting up to test sip lines. We have an unusual router (peplink) and multiple redundant internet connections. We have spend many hours trying to setup our router to enable SIP connectivity however without success. We are looking for someone with 3CX, SIP and good networking /router skills. Hourly rate to be discussed
I need A proffessional programmer who can buil a program that can export members from your competitors Telegram channel and add members to your channels.
Hi Kristen H.,We would like to hire you to prepare a new catalog of @ 170 produc...prepare a new catalog of @ 170 products currently on a GSA contract to mirror and add to GSA Advantage through the SIP database program. All files will be supplied to you. You will need to organize in the correct format and upload both product descriptions and photos.
We need an HTML based SIP client that can be designed to look and act like a in home intercom. For example, there should be buttons for rooms, that will let you page the rooms, and select either video or audio. This will have to be set up that each "station" can be configured which rooms it can page etc... There are a lot more features and customization
App to register with my Asterisk Server as a SIP extension. My Server will send VoIP calls to the App and the App will make a local call on the GSM network and path both calls together. In other words, the Android Phone will act as a VoIP / GSM gateway. Thamk you.
an app that will allow people to join a queue to purchase a product with a countdown and people that are in line will purchase that product when time runs out
...WHMCS version ICT VoIP Billing Panel addition (existing) VoIP Extended Rates Package Rates Extended New* - Enable/Disable toggle for Real-time Billing (just below VoIP Product/Package) (update table) New CDR table to collect all CDRs from [login to view URL] CRON All fields are the same as GetCDRs in [login to view URL] . New column fields UniqueID and Cost
Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own extensions and modify their call routing. I need a developer that is an expert in Node.js as well as PHP because this project will be developed using both languages. If you have strong experience in both languages please contact me with
1. All payments need to go into "project name events" as we need to pay for their meals with this money after the event 2. Adults is £10 and children is £5 3. We also need a tick box asking them of their dietary requirements plus if they would like a meal or not if yes then dietary if not then can we please ask them if we can use it for gaza hospital
Hello, i want script to Test sip accounts with Back SIP response codes Example : HOST = '[login to view URL]' SIP_PORT = 5060 LOCAL_IP = '[login to view URL]' PROTOCOL = 'UDP' USER = '509' PASS = '509123' and it will return me with [login to view URL] (200 OK or 301 Moved Permanently OR 401 Unauthorized etc...) +save
Hello, i want script to Test sip accounts with Back SIP response codes [login to view URL] I will provide : Server ip : Port : Tcp/Udb : Username : Password : and it will return me with 200 OK or 301 Moved Permanently OR 401 Unauthorized etc... +save output into text file +Be able to run in multi-thread Job urgent
hello, i have this package : [login to view URL] need to install on my server windows then build api requests to manage users and add sip accounts SO i will be able later to use on my custom cms
I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio to google speech api (file or stream) 3. get the recognized text back Silence should be detected to stop the file recording or the stream to google Websockets might be used as well
VLC server with the ability to stream locally, installed and tested FreePBX with the ability to connect 2 princess phones locally using either MGCP or SIP, installed and tested I have intermediate Linux knowledge and can assist
hi ,i am looking for a android devloper who can help me with opensource sdk for sip client like linphone , csipsimple [login to view URL] bid if you have experience with sip app , i do not use microsoft products so bid if you are experience in working with linux [login to view URL] will be long term project if satisfied with the [login to view URL] budget is 100$ for this project
Customize microsip: Currently microsip allows parameters to be passed via command line Eg: [login to view URL] /hangupall [login to view URL] /answer [login to view URL] 3892014 (...) You must - change the format of the arguments, that will be passed in the format below: [login to view URL] msip:hangupall [login to view URL] msip:answer [login to view URL] msip:38192014 - add new methods that can...
~50 people across 3 offices (Sydney, Hong Kong - new at WeWork, China) Looking at...(Sydney, Hong Kong - new at WeWork, China) Looking at setup cloud solution to connect existing cisco/ gransdstream sip phones. Currently using Faktortel from Australia - looking at Freeswitch/ asterisk + Twilio SIP trunk (HK phase 1) + Faktortel sip trunk (AU phase 2).
Hi kristenhutchiso8, I noticed your profile and would like to offer you my project. We would like to hire you to set up a new catalog upload to SIP with existing iProd iPrice and iPhoto database files to include @ 160 products on GSA Advantage with our new GSA contract. This is the first of 4 projects with GSA Advantage and FEDMALL we would like to
I have been searching, and can not seem to find a plugin for wordpress to allow payments/transfers via any cryptocurrency, or processing via any blockchain backend (Bitcoin, ethereum based currencies) etc. I would like to build this for a couple of our projects, as well as to sell in the app store. We have our own utility token ready for it as well
...like for you to do is: 1. Configure a Linux Instance (Please provide the build and we can organise it with AWS) to: a. Connects to a Client SIP Trunk (We will use a Hosted PABX to test) b. Connects to a Carrier SIP Trunk (We will use a Hosted PABX to test) c. Passes calls between the Client and the Carrier d. Will hide the IP of both Media and Signalling
I am looki...connections (Upstream Providers) and 1 x PABX / Opensips / downstream. Initial network configuration is completed. Configuration is required for the above + basic call routing and SIP headers. with the requirement for a basic configuration document (outlining works completed) Initial configuration could lead to additional future works.
I need an auotmated software that allows my website backoffice dashboard an ability to drag and drop video content into clickfunnel templates that we upload to the back office. This will allow our client the ability to upload new video capture pages on demand without having to create new clickfunnels.
i have set up a new freepbx server it is online but i cannot get the SIP registration to go through and actually connect to the provider this is not a big project i am missing a setting somewhere and i need someone to connect to my server via team viewer and fix it for me
Need the Integration of Yeaster S100 with Zoho CRM for 32 Sip Accounts and 10 Lines
...outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions: 1. Perform NAT traversal on any incoming or outgoing SIP connections using the "nathelper" module 2. Simply relay the SIP connection to an EXISTING PBX server (3CX). ***Kamailio should relay ALL SIP requests
Looking to have a application that registers to a sip server based on sip 2.0 and or direct ip calling have the ability to make calls. open source can be used. up to 10 favorite contacts can be created in tile format. does not need to incorportate with native android phone contacts(for now) each contact(favorite should have a place to put in rtsp string
...The Gateway accepts SIP VoIP and then communicates with a chatbot over a JSON/REST interface. The Gateway also communicates with Google over gRPC for Speech to Text and Amazon Poly over JSON/REST for Text to Speech. Interfaces SIP Interface - SIP signalling interface. RTP Interface - G.711 and G.722 RTP media interface. JSON/REST API - Chatbot textual
I would like to use the uvc camera on my linphone app to make video connections. If the ...would like to use the uvc camera on my linphone app to make video connections. If the linphone app gives you the source that is normally built, register it according to the sip information I provided and make the video connection possible through the uvc camera.
I AM LOOKING FOR A DEVELOPER WHO CAN FOLLOW INSTRUCTIONS AND WHO WOULD DO THIS AS A PROJECT BUT MUST BE WILLING TO WORK FULLTIME IF NEEDED WHEN PRODUCT IS RUNNING LIVE. This project would NOT be paid by milestonesn because 99.99% of the project is 0 to me and GARBAGE. So if you are one of those scammers looking for money upfront, stop reading now
We need a solution in asterisk that help us block incoming calls with the same numbers (robot calls). We are starting a wholesale VoIP business, but we have some clients that if we have 8 or let say 24 ports open, some times they will send 8 or 24 calls depending on the ports available and all ports will be ringing at the same time with the same
Hello, I need some copywriting - a complete re-write of the below please. You can also incorporate into it these phrases - 'Art uncorked' and 'Art so now'. Have fun creating your very own masterpiece painting with acrylic on a canvas without ever having painted before! Our artists paint along with you demonstrating step-by-step from start-to-end