Hello Our current development website is https://didicar.ca. We need to add Civic plugin to the website. [login to view URL] We tried to install several times, but cannot install and found some php errors. If you have experience with civic plugin, that will be plus. If this project going well, there will be more
Design a logo "PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a..."PaintnSip" We are an art studio teaching art classes for beginners how to paint. Our students are adults and we provide a glass of wine to each student, so we call it paint and sip. We are looking for a fun logo.
...-Search CDR - with the phone number i can see a record when its called and how many time its called by the dialer. -Download Recordings - I can download here records from the server. -Recordings - i can listen here the whole conversation and even edit the resultcode of this client (what i mean with edit is example: From sale i can set it to Negative or
Background: Many times we look for a file on a server like FileExists('COMPUServermyDoc[login to view URL]'), now if the user has rights to that share only via '[login to view URL]ServermyDoc[login to view URL]' then we won't be able to find the file or be blocked from that directory. Technically I can have my program search for the...
I am going to developer iOS app from delphi android source, I am having Delphi android source now, you should collect backend apis with postman style from that. if you were familiar about iOS development,maybe you can work in iOS development stage too. thanks
I have a working FreePBX server. The freepbx server is running good so far. The following changes needed in my server. 1. I want to install a open source predictive Dialer in my freepbx. I have have chosen VICIDIAL for that. You may suggest better one. The dialer must use existing extensions for auto dialing features. 2. You must configure the system
...another VoIP provider. I gave the green color for your easier to understand. -- Executing [s@macro-user-callerid:37] Set("SIP/8902050098-00000024", "CALLERID(number)=8902050098") in new stack -- Executing [s@macro-user-callerid:38] Set("SIP/8902050098-00000024", "CALLERID(name)=+918820094576") in new stack [login to v...
...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
Hi Kristen H.,We would like to hire you to 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 have a Delphi desktop application that sends delivery data to the WorkWave Route Manager web service. Most of the API calls and app objects have been completed. Several additional API calls for retrieving the data from WorkWave need to be completed, where the JSON objects are properly parsed. The supporting objects have been created, but the API
...Our client needs someone who is familiar with Delphi. The existing programs are written in either Delphi 7 (Fixed Asset Pro) and Delphi 2007 (all others). They are all prepackaged, user-downloaded and installed Windows desktop programs that run on Windows 7, 8, and 10 desktops and Windows Terminal Server, Win Remote Desktop Host and Citrix servers.
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.
This job is to translate a small function written in C to Pascal/Delphi. The function takes a string and encrypted it with a seed. [login to view URL] This is an example of how this function works. Find the c program attached in ctcrypt.c , I'm also including a prg file you can use to test to make sure you have it
I have sample demo android studio and want to convert the demo sample to firemonkey delphi (android)
Please make the following changes to the existing code of the SuperScanner you created before: 1) Ensure the entire code works with Delphi 2010. 2) Ensure the entire UI supports different Windows font sizes. E.g. if user is using large fonts in Windows, the UI also has larger fonts yet nothing in the UI must overlap or texts go behind or over any other
...Prevost, Quebec) Tax Rule rate = 14.98% 2min outbound call on SIP Canuk 200 plan 0.020 = 6sec increment 120sec 2min * 0 .020 * 1.1498 = 0.045992 (round up to 0.045) 3min inbound call on SIP Canuk 200 plan 0.025 = 6sec increment 180sec 3min * 0 .025 * 1.1498 = 0.086235 (round up to 0.086) SIP Canuk 200 Package Detail [login to view URL] & [login to view URL] (already e...
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
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
...10.2.2 Delphi Step 1 = - For iOS : add/write/restore* a photo or a video to the photo library of the phone. Photo and video you can take from resource or from the app's work folder. ( as i explain in attached project sample ( project1) ) Step 2 = - Backup and restore* calendar to file, for iOS and Android made in RAD Studio 10.2.2 Delphi Step 3
I have been playing with a simple MP3 player written in Delphi 10 with Firemonkey for use on my Android phone. I hope to release it as freeware in the App Store at some point. I have 3 features to add to this project. Looking for 1 or 2 developers to help add these. 1) Detect when a phone call comes in and pause the mp3 player during the call. Detect
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 need you to develop a software that can store multi fingerprint for same user in a MySQL database and can verify 1 to 1 ( for example as a login ) using Delphi XE2 and U.are.U 4500 digital persona ( source code is needed ) . thanks
This job is by invitation only don't bid unless invited I will send you a link in chat to bitbucket job - I have made the UI I believe it is self-understood and simple. bid for $20 This already connected to my mLab account Real Simple - I had a hard time to construct the JSON doc to use in .update() etc
...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 e. Server will be built on AWS 2. Assist and provide basic training /
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 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
I have a 3CX server hosted on Azure with a public IP address. However, the client IP phones are behind NAT and common NAT traversal techniques such as STUN and TURN are cannot be used. I am configuring an outbound SIP proxy server using Kamailio. There is no database or authentication required. The Kamailio server should perform the following functions: