Asterisk a2billing freepbx jobs
Hello I NEED CONNECTED MODEM 4G HUAWEI for Asterisk AND USE IT IN VOICE Now i use only 3g chan_dongle i need 4G
We are looking for solution like a traditional gsm or CDMA voip gateway. This project will be separated in two is mobile applicatio...application will register to a server and accept call from that server with IXA OR SIP protocol. After that call terminate to GSM Network. (this part like traditional gsm gateway). This mobile application will work only wifi internet connectivity. Beacuse gsm intenret date normally disbale during any gsm call. All call will pass through gsm 729 codec. The registration server may be voip switch or asterisk or any other server. The server will receive call from another voip switch server with sip protocol. Certain number of registered mobile will be able to assign in a group of gateway. On server have have include option show balance through ussd...
We have the basic knowledge of Asterisk and can setup the PBX to make outgoing calls using the PRI Lines, click 2 call etc. Now we need to setup a portal, like Knowlarity / Myoperator etc where we will sell this as a service. a) Sell a IVR (Welcome to our company, Press 1 for sales, Press 2 for support etc...). Asterisk and web development Knowledge is a requirement for this project. b). Once a call comes in its routed as per the rules required and a outgoing call is made to the users number and both the calls are patched. meaning for every call incoming there will be 2 calls. one incoming and one outgoing to the users mobile and both are put into conference and are recorded. c) similarly we could make a outgoing call using the board number by calling the board number and th...
We have a FreePBX installation. Our customer has a CRM used by call centre consultants and hosted at a different cloud provider. All telephony based activity is done via the CRM (no physical phones or 3rd party soft phones) to connect to the FreePBX server. The customer has a implementation, using WebRTC. The implementation is making use of 2 secure connections (wss://); 1. to handle the voice and SIP 2. to handle server requests such as Login, make call etc. and receive call progress data such as channel added / removed, connected An important feature for the call centre is that all sales calls are starting as a conference, this to create a customer experience that allows for adding additional persons without the unpleasant silence etc. associated with being put into a
No companies, individuals only Your profile: Young, talented, efficient, you are looking for a challenge. We are a remote team working on interesting innovation projects and are passionate about our job. We enjoy what we do and we do not count our hours but we measur...: 1st interaction: "register user" Ask what language would he like to speak. In the language of choice: Ask for birthday Ask for Name Ask for his address 2nd call : Tell the user what you know about him Hey {name} you are currently {age} in the right language. what would you like to know? User can ask: What's my name? What's my age? What's my address? OPTIONAL Angular React Ionic Openvidu Asterisk Sails Web socket php symfony You are expected to work in GMT+2 timezone. What budget do ...
Hi guys. I'm searching for an experimented person on Patton SN4171 e1 gateway configuration including trunking with ISSABEL IPBX (Asterisk based) and call routing configuration at the patton side.
No companies, individuals only Your pr...successful with you and grow together! We are looking for an agile developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different techologies by self learning. We are looking for attention to detail, communication and reliability. Mandatory expertise Nodejs Angular Sails Websocket Mongo OPTIONAL React Ionic Openvidu Asterisk Chatbot php symfony You are expected to work in GMT+2 timezone. We pay 2500 CHF per month and offer 20 days of paid holidays per year. TO APPLY 1. Let us know your experience in years and short description of your work performed on all mandatory elements and ideally on optional elements. 2. Send your CV 3. Put your bid for monthly in...
hello i have a raspberry pi that have Rasterisk on it i am using chan_dongle solution to use Huawei E160 for a GSM port my end is this : Port1 is talking B, during this time a sip call(from C to D) is coming to Asterisk , Asterisk should hold call on port1 and B , and port1 should call D when call connected to D , port1 should merge all calls (Conference) it means now port , B , D , C is on one call
I am looking for someone to install software link below including with all dependencies e.g: asterisk, soundcard all installation process is explained just need someone who did this past. Please with no experencie or familier please dont bid,
We want to integrate asterisk free PBX with Avaya and siemens PBX for one of our customer... If any one local KSA available let us know..
The task is to search a VSTS repository for exact and fuzzy keywords in the source code files in the repo. For example, if searching for the keyword "ADD", the tool needs to scan all the files for "ADD" as well as "*ADD*" where the asterisk can be anything (or nothing), but not including spaces. ASCII characters only in the keyword list and the VSTS repo. Case is ignored, so “ADD” should match “add” and “aDd”. The keywords will be in an ASCII file and need to be read from top to bottom, then each file in the VSTS repo scanned for an exact match of the keyword, then for fuzzy matches. The tool can scan all exact matches first, then all fuzzy matches, or it can scan for the exact then fuzzy match one after another,...
We need to install asterisk to the debian 10 server. We need to start the asterisk on server, add 2 SIP accounts and organize SIP calls between them.
I need a team of developers with speciality at developing Asterisk based call taking applications , I also need a Computer Aided Dispatcher Application Developer with speciality to implement GIS based solutions.
... Tables and Figures should be centred, and numbered and titled in bold capitals. Acknowledgements should appear at the end of the main text and before the references. Any appendices should appear before the References. Footnotes should be numbered consecutively in the text and placed on a separate sheet of the manuscript. Any footnote attached to the main heading should be designated by an asterisk. References follow the author-date Harvard style. References in the text should give the author’s surname, year of publication and page number if a direct quote is included. References should be listed alphabetically after the text. Journal and book titles should be written in full. Some examples are: Baumol, W.J. (1986) Microtheory: Applications and Origins, Cambridge, MA: ...
Looking for Asterisk PBX Developers who can do deep customization and integration to various in-house applications.
I have a working Jitsi-meet server, and a working (production) Asterisk Server. Would like to be able to dial into a converence. Have installed jagasi and it registers with my asterisk server. But stuck from there!
kindly be informed that we need to do some customization for issabel call center module which is allowing logged agents to do outbound calls manually, please be informed that we want those manual outbound calls to appear on call center reports and recordings Skills: Linux, PHP, Software Architecture, VoIP, Asterisk PBX
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
I am looking for a developer to develop a telecom system and android and iOS apps.
We have an asterisk server at our HQ and we have sip phones in our branch offices. We want to Develop a SIP softphone that we customize with our brand and users can install in their windows, mac, ipad, android etc....We need to do voice and video calls with the softphone.....something like LinePhone and Zoiper.
1). Install 4 Asterisk instances Time for completion : 2 days 2). install and setup Kamailio on another isolated VPS instance; Kamailio will be setup as the registration server for all clients, and it will load balance all call requests on each registered Asterisk instance. Time for completion : 1 day 3). In conjunction with app developer, configure all the necessary dial-plans and trunks on the Asterisk instances and make sure they work with the database and scripts created by the developer to compute predefined data, and setup Asterisk to perform actions based on the computed results. Time for completion : When needed 4). Provide support in configuring SPA3102 devices (PSTN gateway/converter) Time for completion : When needed
I am looking into having my own miss call server/tools. I require assistance to: 1. Create / rent a server that meets the requirements needed to support sending miss calls to number from a local database. 2. Have a tool that sends ( dialer ) the outgoing calls 3. Setting up to be able to receive calls back from the numbers ...be able to receive calls back from the numbers called and redirect them to IVR or CC 4. Make a report at any point in time with current costs / income. I already have suppliers and trunks to connect to. Already have databases. I also accept consultancy to guide me in buying tools already on the market and putting everything together in a working manner. ( like what server to rent, installing asterisk PBX, setting up trunks, dialer, database which i can buy s...
Coding tips and live Double verification and more..
hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks
hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks
hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks
hello I would like Trixbox / Asterisk installed on a VPS, and have trixbox working and to put the install steps on a text file for me. Thanks
I have a system asterisk 3g modem, I need to install for me a database for IMEI of dongle purpose for this operation is when i put a dongle with new sim in my system dongle change automatically it imei. Pkease confirm it is possible to do it.
...Freepbx's Zulu Mobile client (iphone/android). Right now the issue is: On a new server, I can use Zulu's PC client to do outbound/inbound to external calls, but with the mobile client, I always see this error below when I do an outbound external call, and the call will hang up itself, the mobile client out close it by itself too. I've tried this on both iphone and android client. Can any asterisk/freepbx professional provide me a quick solution for a fee? Full log: -- SIP/6566811234-00000005 answered PJSIP/90101-00000005 -- Channel SIP/6566811234-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26ff-48b1-aece-77a2e185c45e> -- Channel PJSIP/90101-00000005 joined 'simple_bridge' basic-bridge <3de645f6-26f...
Our main goal to minimize the BW in client side with good quality of voice . We need some kind of bandwidth compression system ( upto 60-80% than usual SIP calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls from server A and sending to gateways (quintum gateway for example) or E1 cards. 3. Number of Server B can be unlimited. 4. Number of Gateways/E1 cards per server B can be unlimited 5. For server B installation need easy to use ISO image that could be booted from U...
... I need to replicate there backend using only asterisk. That requires implementing asterisk to my AWS server. To be specific , I need a developer to add asterisk to my server and develop a secure rest api that will prank call two people for you using asterisk in my server. Please use the website i linked above to understand exactly what kind of prank calling I'm trying to implement. After the call is made through the server it should record the call up to 30 seconds , 1 min or 2 min only and play and audio file in the end before the call ends. This audio file will be provided by me. This audio file will be saved via my google firebase could storage. After implementing asterisk and making the functionality work. I need a secure rest api to call ...
We are looking for a passionate and talented VoIP Engineer to join a multicultural team (including French, Russian and Romanian) and help us tackle ambitious telecom challenges. You would be part of a growing tech team of 30+ people that builds a cutting-edge international VoIP network on which relies our SaaS customer interaction management solution. This solution all...innovative person by nature, you are eager to learn new technologies and to take up new challenges. You master most of the following technologies and methods : * VoIP protocols (SIP, RTP, WebRTC) * Analysis and investigation (logs, IP frames, pcap traces...) * Shell and Python scripting * Automation (ansible) * Monitoring (collectd, grafana, nagios) Bonus: knowledge of Oracle Acme Packet SBC, Asterisk, FreeSwitch...
I want a asterisk call base caller/callee filter solution on PHP which can filter incoming traffic and create whitelist and black list base on condition of repeated callee and non repeated callee with voipcall bandwidth optimizer..It will also work from 1ip to another ip.. Blacklists and whitelists of phone numbers/ Caller IDs on the basis of repeated numbers. Maintaining Call Gaps between calls to simulate HB Monitoring number/Caller ID length pattern while receiving calls Optimizing Bandwidth Efficiently
I have a raspberry pi with a working asterisk 16 and chan_dongle I am looking for someone who can 1 capture dongle data from ami and update in db 2 assign static names to each port on my usb hub so that port 2 will always be port 2 3 insert dongle imei in a mysql table based on which ports the dongle is in 4. call a file if there has been an imei change on a usb port
I have just installed Fusion PBX and want to have my Polycom phone auto provision with TLS encryption using ZTP config file. I already have a working ZTP config file that works with my FreePBX server using TLS, but I cannot get it to register to my Fusion PBX server and I think its the settings of the Fusion PBX. I need someone who knows how this is done correctly. I DO NOT WANT ANYONE TO TRY TO FIX IT, I NEED SOMEONE WHO KNOWS HOW TO FIX THIS. THIS WILL NEED TO BE DONE IN ONE HOUR FROM START.
I have just installed Fusion PBX and want to have my Polycom phone auto provision with TLS encryption using ZTP config file. I already have a working ZTP config file that works with my FreePBX server using TLS, but I cannot get it to register to my Fusion PBX server and I think its the settings of the Fusion PBX. I need someone who knows how this is done correctly. I DO NOT WANT ANYONE TO TRY TO FIX IT, I NEED SOMEONE WHO KNOWS HOW TO FIX THIS.
Hi, I'm building a website with Gatsby. I need someone to fix an issue on the localization part, some pages work some don't. No agencies please. In your proposal, add three asterisk at the end.
I would like to integrate Exotel system (based on OpenVBX) into my FreePBX server for seamless inbound and outbound calls. Freelancer must have experience to integrate the APIs from httpsdeveloper[dot]exotel[dot]com/api/ as a Trunk and utilize other applets from httpsdeveloper[dot]exotel[dot]com/applet for managing calls to ensure the combined system effectively helps to stay in touch with our customers at anytime, anywhere.
I want a new asterisk server built to replace our old version. i want someone who will build the server from scratch. Not just copy the sample files and fiddle with them. i require the following functionality. 1. Twenty Extensions. SIP Phone and Softphones etc 2. Conference bridge with IVR to Choose a room. Must support Video and Messages. 3. Short text message between extensions. 4. Video Calling between extensions. 5. IVR with Time of day manu for office and none office. Dial by name eyc 6. voicemail for all users. 7. outbound dial for two factor authentication. needs to call end user. ask question and receive response. (details from database ) 8. 4 Analogue Lines. Digium Card installed 9. 2 SIP trunks. Linked to group of extensions. 10. hunt groups. 11. Connection to Mic...
...developer with skills in multiple technologies including Node.js and angular, ideally some php as well with the ability to adapt to different techologies by self learning. We are looking for attention to detail, communication and reliability. If you believe you are a good match for our team, please let us know your thoughts on the following challenge. If you had to create a voicebot integrated asterisk in 2 languages with a voicebot needs to do the following: 1st call : register user Ask what language would he like to speak. In the language of choice: Ask for birthday Ask for Name Ask for his address 2nd call : Tell the user what you know about him Hey {name} you are currently {age} in the right language. what would you like to know? User can ask: What's my name Wh...
Hello, the script already exits. There is a documentation. I need a full integration of the script in Node.js, one side FreePBX and the other Amazon Lex (or google speech). Possible upgrade with UniMRCP. here is the link here is the explanation on how to configure
Hello, the script already exits. There is a documentation. I need a full integration of the script in Node.js, one side FreePBX and the other Amazon Lex (or google speech). Possible upgrade with UniMRCP. here is the link
Hello, we created the phonebridge, now there are some problems to fix: 1) When the call starts on zoho the pop up shows alway "in connection". From logs we can see that - On RINGING state, dialing channel is null - On UP state, linked channel is null - on HUNGUP state, dialed channel is null 2) When the softphone rings (inbound call) on zoho do...created the phonebridge, now there are some problems to fix: 1) When the call starts on zoho the pop up shows alway "in connection". From logs we can see that - On RINGING state, dialing channel is null - On UP state, linked channel is null - on HUNGUP state, dialed channel is null 2) When the softphone rings (inbound call) on zoho doesn t appear any pop up Will be provided a vpn to access Asterisk pbx and admin l...
We are looking to own an open source(preferably) SIP media server and stream audio on it. This has to receive audio stream from asterisk. This has to start automatically when conference starts up on asterisk (our pbx). We have the PBX'es, just need that audio stream / internet radio that can be accessed by those who can't call in.
hello I hope you all are doing well in this period. I currently have a a2billing system with asterisk, I would like to move to a2billing, kamalio with all the benefits of it and eventually integrate asterisk if necessary. Budget is tight as well as time schedule but there is a great opportunity also for the maintanance phase.
...conference rooms, and webinar (named presenter mode) options. Both services provide apps for android and IOS, and use webrtc as the base platform for delivery. We require a php front end that connects to Jitsi and provides all the features of zoom. Features needed Connection by phone (SIP trunk) This will be a SIP number that terminates on asterisk (freepbx) IVR which will request conference room number and pin based on the information (numeric only) provided by the script. This will require using asterisks API or .call file. Interface Main interface for users should be controlled by Username and password User should be able to create conferences, schedule by time, and so on. Create a free account on Zoom for live examples. Once a conference
I am looking who understand voip scanning . basically I am looking to install following program from start : please read and let me know if you can do this for me ?
