...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
We've an operational GLUU-Server and OXD-Component connected to one Wordpress-Site. Now we need to add functionailty to some Apps in order to use the same Oauth-mechanism. Also Signup-Functionality needs to be setup and customized, so that new users can easily signup for GLUU-Server. Later we'll need to add Authentication of users against other OpenID-Providers via GLUU. When you appl...
Integrate PBX ( Cisco, Avaya ) with Middleware for Phone Lock/Unlock, Assign Name to extension, create Voice Mail, Set/Delete Wakeup Call etc. More info will be shared when the candidate is selected.
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.
Hello, We need to add a functionality to our IVR which is based on Asterisk V 13.14.0 / PhpAGI. Os is Debian 8, database is MySql 5, Php is also 5. Simple functionality: - Inbound call accepted (client who needs support) - IVR (PhpAGI) says "welcome" - Call is forwarded to 1st level agent (already done by DIAL command) - 1st level agent takes call
...We offer: - hourly wage: 15 USD/hour; - wages minimum 10000 UAH per month; - work in a prospective company: we introduce automation systems based on open-source products Asterisk IP-PBX and CRM VTiger: open API, easy integration with other systems, more than 30 own developments for CRM VTiger, VTiger has wikipedia and community community developers;
...(landline 1) and I have my own server too. I'd like to redirect calls made to landline 1 to another landline (landline 2) that I don't own. I've already made a vocal robot on Asterisk so that depending on the digit the caller presses, it does something different. What I need now is the last part : 1) Depending on the digit pressed, forward the call to a
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.
I am with a security company in Anaheim, and need a tech to configure (25) 2960's and (1) 5672UP. Ip addresses and VLAN mostly and a test that all the devices can be seen. There will also be 54 Meraki wireless devices connected to the system as well
...require someone hands on experience in installing goautodial or vicidial solution on Google cloud computing. Good understanding of Linux, Asterisk and Vicidial is essential. We wish to start testing Asterisk and Vicidial in the cloud from Google. We require setup, testing and ongoing support. Phase one is setup so we can start testing. Phase 2 will
I use open source called linphone I have implemented the sip Android client. But it is exposed black screen. It seems to be losing packets. So I want to implement a separate android sip client. My SIP server is implemented as Asterisk. My requirements are as follows 1. Registering with the Sip server - register with id, pw, ip, port, expired time, transport
...outsource for those tasks when needed. Currently i can be specific on a project which you can tell me you can help me on this or not. We have some raspberry pi products which asterisk is installed. And there are 3G or 4G usb modems on them. Those asterisks receive calls with IAX trunking and route calls to mobile phones which are matched with raspberry
zoho's phonebridge does not fully support asterisk 13, probably because of java issue causing the plugin not to send the correct information to zoho i need to be able to get full feature set from the phone bridge : caller id + popup + call duration once answered click2call + call duration once answered all call data should be listed in the crm based
I am having problems getting a dhadi/Asterisk/POTS configuration ( based on an old Zaptel/Asterisk configuration that does work ) to work properly. Config files and CLI output: [login to view URL] Dialplan: [login to view URL] The machine once finished will backup and replace my old Asterisk servers that basically operate as POTS
This is simple task to find good web ...pspdfkit example source code, I need to integrate it to my [login to view URL] project using webpack. [login to view URL] Need expert of webpack configuration, you should know about [login to view URL] and npm. Budget is $30. I am going to give other tasks if you give me good result for this project.
I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer • Blind Transfer • Supervised Transfer •...
...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...
...backups. I also need to set it up with a Control panel like Plesk or cPanel. It needs to be setup with a hosting server, mail server, SSL for domains, and I will set up an Asterisk server in one VPS which also needs to be secured. I also need help migrating over from my current VPS to the new dedicated server setup. If you think you can do this please
Hey everyone, I'm working on a project to develop an interface for a VoIP server to allow users to add their own ...languages please contact me with a proposal and make sure to write "iluitech" in your message so I know you actually read the requirements and know that you can do it. If you have SIP experience and know PHPAGI it would be a huge plus!
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...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
Need to develop a code pipeline workflow that can combine builds from two different repositories into a single build artifact to be deployed to Elastic Beanstalk. See attached image for workflow that is desired. Frontend application is angular (source is in client) -> builds into a folder called "client/dist" Backend application is nodejs (source is in server) -> builds into a f...
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 setup cl...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
...jams * Wheel and wheels wells cleaned * Exterior windows * Tire shine Add-on * Spray way $50 * Leather clean and condition $50 * Rubber mats $10 * Trunk vac $5-$10 * Engine bay clean $20 * Floor mats shampoo $10-$15 * Carpet shampoo $30 * Door panels cleanded and conditioned $20 or $5 each * Full interior $150-$200
I have an issue to configure docker and ipv6. The following need to be archive I have /64 subnet I need to be able to use in docker. I want my container to run on a static ipv6 address. I try to run multiple containers with different ipv6 on the same port.
...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