Elastix pbx asterisk pbx işler
my provider's server crashed (raid issues) and i need a find gentleman who can install freebpx / asterisk and secure it on a cloud server having ssh access...
...we would like to implement dialers and outbound stuff but in a more basic way, not yet to implement predictive or so. Also we would like to have a Webrtc phone integrated in the agent panel that they can interact with. So far we are developing the tool using Laravel, Vue.js and MySQL, the application interconnects with FreePBX to get data and eventually the idea is also use the capabilities of Asterisk/FreePBX through AMI and AGI to trigger actions like make, receive and monitor calls. I would like to see the possibility to talk to you and see if you would be interested to help us with your knowledge and experience in the VoIP related area. We are looking to give to our customers a really nice tool to use and we are open to suggestions if that would be the case and make this pro...
Hi We have installed vital pbx 3.0 and we are looking at moving all our clients over from kiero operator to vital pbx. I'm looking for someone to move the clients over one by one. I will advise you the call flow setup along with a zip file for each client for IRV's, welcome messages, hold music, etc You are then to setup each client as a Tennant in vital pbx Along with configuring other vital pbx settings for clients to get the most out of the platform. We as a business will be the first customer to go live on the new pbx. Once we test the platform for a week then other clients trunks can be added and they can go live We have around 100 extention across all clients With your experiance we will also look at other wholesale sip provid...
I have an asterisk server, the calls are droping as soon as the call reaches the IVR, I require help to fix this problem urgently please.
We have a Asterisk/FreePBX-system with 2000 preconfigured extensions. We need a software to run on another Linux machine that continuously monitors the status of the Asterisk/FreePBX system state and triggers email-alerts if the server is unresponsive. The software also need to log in a local SQL DB when a PJSIP-extension/endpoint on the Aserisk/FreePBX changes state. Details of the Endpoint should be logged with the state change and email alerts should be able to be configured on specific extensions and state changes. (I.e Unavailable->Available/"Not in use" and Available/"Not in use"->Unavailable). The software needs to send heartbeat email alerts that it's running and have access to the Asterisk/FreePBX server. All email alerts need t...
We have a single Virtual Phone Number that we want to direct to our Linux cloud server, most likely running Asterisk. We only need a single account so that when the Virtual Number is called then will be forwarded likely via SIP to the servers IP. This will then play a single Audio file of our choosing. Perhaps later we might expand it into a full auto attendant, but we only really need a single message then the user will be forced hangup.
our Gsm Gateway software use PCI gsm modules board as its Hardware we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base
Im getting hacked , to the point its slowing down system I need someone to stop this
**To do without any commercial module** I have a requirement of Asterisk/ FreePBx to make it redundant. for that what my client is looking for is One main PBX at Location A and another PBX box at Location B. In case Location A goes down for any reason then Location B should kick in and continue the phone service for the company. Is this is something you can do? Please let me know.
Required expert in Cisco phones to setup custom features for the phones. We are using Elastix pbx server. Would like to configure phones to auto pickup on speaker phone without ringing.
For custom made Invoices to bill clients for incoming calls through our Asterisk PBX or for items sold and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc spreadsheets 2) Software to change templates to pdf without loosing formulas and calculations (Foxit ?, etc.) 3) Linux 4) MySQL DB 5) PDF 6) PHP 7) KVM qcow2 (if possible, for disk management) We use background images in LO Calc spreadsheet, to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image ! This project is not for beginners !! You must have a PC with a Linux OS (Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This ...
We are looking for a Call Filter solution for Asterisk .Call Filter will Whitelist and Blacklist numbers using rules.
I have an asterisk server which has oem set functionality which I want to be accessible via an android app. Below are the requirements. Allow users to be able to call each other within the netowrk via UserID - users must be able to call each other at anytime i.e. constantly connected to the SIP Include speech synthesis in the app - so user can type in what they want said at the start of the call and it gets sent to server Ability to set Caller ID internationally View call history Set voice pitch dynamically through app if possible Receive DTMF dial tones as sent to server by callee Must be modern design and run smoothly. Must include FAQ with details on how to use the app Must include help page with contact information On startup: must include a progress bar with audio file play...
Create a JavaScript program that will calculate the volume of the cylinder. The formula needed to calculate the cylinder volume is: V equals r squared asterisk times pi asterisk times H The height (H) and the radius (r) are arbitrary, while Pi must be defined as constant (3.14). Record the result using the alert method or document.write. Note: Send the work as an html (zip) file.
Hi Mohammad Saydul K., I noticed your profile and would like to offer you my project. We can discuss any details over chat. The project is about to integrate Teams, Kamailio and a SIP provider and asterisk. Teams and SIP provider registration is already working, including certificatates etc. The major part of the work is to configure Kamailio to rewrite phone numbers and make RTP, sRTP work.
Need an autodial script which can work effectivily with free pbx server .
Hello, Need to setup audiocodes with asterisk, 4 analog lines and route to trunks. Work is via Anydesk or teamviewer. Thanks in advance!
We are a managed voice provider supplying telephony to businesses and call centers, we require a developer to build us a license based (we intend on licensing and reselling to clients on a subscription base) reporting and analytics add on tool for the PBX system in question. More detail will be provided in initial negotiations.
We are looking for an individual or team to polish our current app and fix bugs. It is a Chat App with SMS. Qualified candidates will be able to review our list of bugs and create estimate base on that. THIS IS AN ON-GOING project ...candidates will be able to review our list of bugs and create estimate base on that. THIS IS AN ON-GOING project and we will need you indefinitely. So please apply only if you have the time availability. Skills required: Front End: Angular, JS, Bootstrap, JQuery, CSS, HTML Back End: PHP Framework: CodeIgniter Web Framework NodeJS, JS, composer, PHP, JSON, Angular Cli, MySQL, General Requirements: Asterisk, API deployment, SSL Certificates, Networks, CentOS, Unit Testing, SMTP communication, FAX communication scalability experience good documentat...
Set up Kamailio in front of asterisk server to proxy webrtc/tls /tcp / udp to asterisk server on private network with failover. - rate limiting sip traffic by source IP - dropping malicious / invalid packets - integrated with APIBAN () - MySQL integration to be able to adjust config (like primary / secondary / tertiary failover etc) - auto install script so this can be reinstalled / moved / etc easily with minimum effort. - Bonus for early start / completion within 24 hours
Hello we are looking to install Kazoo pbx system on our server
I need to create a virtual pbx with their respective ivr, queues, extensions and this needs to integrate with zoho crm and zoho desk and additionally I must implement a chatbox from our website and that is integrated with zoho crm the idea is that users can make calls receive calls schedule calls from zoho through the pbx in the cloud configured in twilio
What shapes are needed and how they need to look: strLine: This variable will hold a line of asterisks strRectangle: This variable will hold a rectangle of asterisks strTriangle: This variable will hold a triangle of asterisk. strSquare: This variable will hold a square of asterisk. shapes should be able to change orders in that they appear.
1. Linphone registration a) Li...REGISTER message containing its PUSH token via Kamailio to the PBX b) When Kamailio receives the OK from the PBX regarding the successful registration it stores the Linphone push token and username Incoming INVITE to Linphone PBX---->Kamailio a) Kamailio gets the INVITE from the PBX and does not forward it to Linphone b) Kamailio sends instead a PUSH to Linphone to FCM (google firebase) c) When Linphone receives the PUSH it issues a REGISTER message d) Kamailio catches the current contact address from the REGISTER message e Kamailio now forwards the INVITE from step b) to the current contact address of Linphone f) The usual INVITE from step a) proceeds SIP-INVITE ----->[PBX]----->[Kamailio]push------&...
Looking to build a setup where. 1.) UA /WEBRTC client registers to kamailio 2.) Kamailio is registering directly to PBX for all SIP. extensions. 3.) setup where Kamailio doesn't use RTP engine (not sure if i can) need to test config is almost there Need some assistance/tweaking. as i dont know.
Hi I have CCs Zycoo coocenter-S30 , need many modification for the following : 1. Change GUI for monitor user Interface ( add counter for the queue ) also Agent user (Arabic language) 2. add some math equation in reports (Arabic language) thanks
We have a basic setup ready with chan_dongle and M35 modules. We are able to run this succesfully with Asterisk. At times asteriks crashed intermitantly . Some time the gateway runs for 24 hours without issues. But some time it crashes with in few hours. We need some one with experience on Asterisk and chan_dongle who can resolve the issue for us
We looking for a VoIP / Asterisk PBX professional that is great at sharing knowledge. We working on a VoIP solution that will be integrated in our complex public transport system. Currently we have implemented a simple 1 to 1 voip call but we planning to extend the implementation. The features we want to implement are following: - Group calls (1 to many) - Custom headers for additional configurable options (mute all participants, call priorities) - Call Auto-accept - Calls history - Group calls rejoining We want to extend our VoIP / asterisk pbx knowledge and validate the approaches to some of the features described above. The responsibility of the professional will be preparation of one day virtual session (preferred Microsoft Teams) for 3-4 developers to ...
We are a wholesale distributor that needs a way to input a list of orders and automatically call the customer and play a pre recorded message.
we a Gsm Gateway software that use PCI gsm modules board as its Hardware we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module
Currently we have a gam gateway call management software , the software is working pci boards hardware . our project now is yo change the ( pci boards GSM modules) based interaction with , USB GSM Modules based interaction . The Developer modify our software to work with USB gsm modules , the developer need expertise in VoIP. Asterisk. Asterisk Modules. GSM Modules .GSM Chan , GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers
Use webRTC or native language to develop VOIP dialer & call and messaging app for Android and iOS with the following features: Multi-language support: English, French, and Spanish User Registration: Language selection, OTP / OTC for user registration, etc Phone Contact Integration: Sync phone contacts each time app launches Chat with media:...French, and Spanish User Registration: Language selection, OTP / OTC for user registration, etc Phone Contact Integration: Sync phone contacts each time app launches Chat with media: text, images, emojis, audio, video, documents and other files Group Chats with media Audio Calls Video Calls Group Audio calls Profile management Notification management Message Search Option Star message Settings Asterisk Server integration Push Notification ...
i need filter on asterisk about did and clid number. We are using this project on maniplation temination voip traffic
...to answer a short survey and the call will be transfered to the survey module. The customer will be asked questions and asked to press the appropriate number key to indicate their response. Make IVR surveys form excel phone list. Receive answers from phone keys and voice recognition. Detailed requirements: * Must be delivered as a FreePBX module compatible with latest versions of FreePBX and asterisk * Must be configurable through a web page on the FreePBX web interface * Configurable features must include: number of questions, range of values for response * Results will be stored in a MySQL database with information on the call, the agent, the questions and the answers to the questions * There will be a web interface to display the result for each question across multiple age...
looking for someone to install FREEPBX 15 on Centos7
I have several FreePBX / Grandstream PBXs, hosted and on-premise, I am looking for a monitoring system that can help me with the following: 1. Automate test calls and verify two-way audio 2. Detect packet loss, jitter, or latency issues 3. Monitor SIP and RTP packets 4. Monitor SIP registration for specified extensions / trunks 5. Alerting capabilities 6. Does not overuse pbx cpu/memory
Hi Alexander, y noticed you finish a project named: Free Pbx IVR survey module. I am interested in the same project with some changes. I would need to see screen shots of the project in order to discuss with the responsable of making the software work. It is my understanding that the project uses phone keys to respond the survey, I would also need voice recognition availability to respond the survey. Please let me know if you are interested. Sincerely A. Ayala
...transferred to security. Proof of concept (POC): A mobile app (Android and iOS) to demonstrate your firm’s competency to create a two-way audio/video call from the smartphone to our target Entry Phone. The target uses ARM architecture, runs Pine64, an embedded platform and has Node.js based API running on Linux. Deliverables for the POC: We would like to see a SIP VOIP call placed through our Asterisk server between the Entry Phone and the app. Establish a webRTC to webRTC connection having one-way video and two-way audio streaming. This call will be initiated from the entry phone to the resident’s mobile app where residents can see the visitor and be able to talk to them. Establish a webRTC to webRTC connection having two-way audio VoIP call for offsit...
Hello i have asterisk with did where 500 users call this did number we hava to create a routing logic where we will interconnect them randomly inside dynamic conferance without pin if in the conferance caller or callee press 0 we will interconnect them randomly to any other conferance without pin in case the user have no any user available we will interconnect hem to any active conferance else if there is no other user we will play MOH until the any conferance will press 0 we will set a timeout for the conferance 120 secs this is small job but i will have many others
Would like to connect Zoho CRM with local Israeli accounting software, local Payment GW as well as loal PBX via API.
Would like to connect Zoho CRM with Local israeli accounting software via API as well as Payment Gateway and local PBX provider.
Hi, I'm in need of either a Discord bot, wordpress plugin, or a webpage I can integrate into my current site that users ..."spectate" the stage strike/select process of any given matchup. Lastly, it is very desirable if the system is graphic based. I can provide the images for each stages available, but would be required for mod to update stages available. For now, placeholder text can be used: *Stage 1, *Stage 2, *Stage 3, *Stage 4, *Stage 5, *CStage 1, *CStage 2, *CStage 3, Player 1, Player 2, Room Code, Winner Status (asterisk items will be graphical) Would finally need to know from the individual chosen for this project, which method would be best to produce this in, Discord Bot, wordpress plugin, or web page Willing to work closely to get this done. Looking fo...
Hello, we are a new little italian isp provider company. we are looking to build our own CLOUD PBX SYSTEM with MULTI-TENANT. i hope someone can help us! Regards
we need a developer that should have these experiences: Modules. GSM Modules .GSM Channels, GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers the developer will develop the software will run GSM Gateway and SIM Bank using Asterisk and GSM Modules, Hardware and software knowledge required. payments terms . please read it , will will not release any milestone until job is done , not negotiable
we need a developer that should have these experiences: Modules. GSM Modules .GSM Channels, GSM Gateways, rSAP -> Remote SIM Access Profile, rSAP over TCP,,SIM Banks ,SIM Servers the developer will develop the software will run GSM Gateway and SIM Bank using Asterisk and GSM Modules, Hardware and software knowledge required. payments terms . please read it , will will not release any milestone until job is done , not negotiable
We are looking to support SMS on our MiRTA platform (Asterisk). We also want our clients to be able to utilize SMS using the Zoiper app (open to other options as well). MiRTA is the GUI built on top of Asterisk. We will want the SMS options and features built into the MiRTA GUI for easy deployment and provisioning (we have access to the MiRTA developer). Please provide a time estimate. The budget is suggested and can be adjusted depending on your expert opinion of time. Please provide feedback. Thank you!
For custom made Invoices to bill clients for incoming calls through our Asterisk PBX and Credit Notes to pay telephone agents for their work. Experience needed: 1) LibreOffice Calc spreadsheets 2) Software to change Calc templates to pdf without loosing calculations (Foxit Reader ?) 3) Linux 4) PDF 5) PHP We use background images to show invoice and Credit Note Templates to the user. To fill the Templates out, on our server, the user only sees the image, the spreadsheet is invisible ! This project is not for beginners !! You must have a PC with a Linux OS (pref. Ubuntu). ----- Good and quick communications are essential !! - If You check Your mail only once in a while, You don't qualify. This project will be divided into unequal milestones. See attachment.
I need guidance on how to setup a blank sim card to connect to asterisk server and allow user to set caller ID
