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    10,734 install asterisk server jobs found, pricing in USD

    I need my gateway to be set up so that I can receive international voip traffic and forwarded using the simcards on the gateway. You set the route and Asterisk server on VPS. This should take just few minutes

    $38 (Avg Bid)
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    I am looking to hire someone to link Asterisk with AWS Lex or [login to view URL] and add a Twilio or similar provider. I need an easy interface to update Lex or API.AI. This will include the ability to edit complex dialog flow through a user interface. The ability to route different customer phone numbers to and from different AI flows. When the AI flow does

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    Asterisk development with HTML5 interface. - admin web interface to create corporations/tenant PBX - customer web interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation

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    Hell...single field from out PHP program to a linux/asterisk script/process that will: 1. Take the number as an input value 2. Attempt to call that number provided for 1 second from Asterisk/Trunk 3. Catch the return code of the attempted call 4. Pass the return code back to the calling PHP program We are using Asterisk 13.19.1 and FreePBX [login to view URL]

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    ...calls can me dialed ( Outbound ) Via the GSM ( the SIM card which is installed in the phone ) In more details Now we have a mobile device and Asterisk server, I want to link the mobile to the server and this server will send the call (SIP) and the mobile will make this call via the SIM installed In a way or another this APP will make the mobile work

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    A2billing with asterisk installation Script + Configure the basic calling card and DID function

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    i Need to configure FXO/FXS gw (SPA3000) with asterisk. the interface system is xCALLY. i have connected SPA3000 with my DNS (fixed domain from [login to view URL]) the connection between asterisk server and SPA3000 DNS/IP should be via VPN which already setup on asterisk server i have uploaded 2 image showing the xCALLY trunk steps/inputs.

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    ...sure i can use a SIP account as a routing gateway I want to make same to same VOS3000 softswitch If you want to Use asterisk Use JAVA-AGI database must be PostgreSQL Priority musbe be have to work you can use any good Engine like asterisk,FreeSWITCH,Kamailio,OpenSIPS,etc More info check Attach file here is all screen shoot i need a complete wholesale

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    I have an asterisk server: Asterisk 1.8.20.0 built by palosanto @ [login to view URL] I need a php based script which will popup a browser window for every incoming call & show information based on QUEUE & DNID. Budget: $50

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    ...in Germany. We have multiple Solutions and now are expanding to telecomunications security. We need developers with experience in VoiceXML and JS, as well as knowledge in Asterisk. We are building an IVR system with dynamic and automated functionalities and offer Jobs with contractual regulations for at least 3 1/2 years, after hiring! We pay on a

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    hi duke, i need a team to integrate Asterisk call center with Hubspot to have some feature bellow: - Sale can use Call feature on Hubspot via Asterisk. - Can use feature Click to Call. Please show me the solutions (brief, draft with your paper) to continue!

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    ...dtmf or voice dial commands based on what comes in the sip header. the sip app will register to a sip sever who will send it numbers to be dialed in the header When the sip server sends a call, the sip app will issue dial command via the headphone speaker. (to phone 2) if it hears a acknowledgement that the dial was received, wait if no acknowledgement

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    ...in Germany. We have multiple Solutions and now are expanding to telecomunications security. We need developers with experience in VoiceXML and JS, as well as knowledge in Asterisk. We are building an IVR system with dynamic and automated functionalities and offer Jobs with contractual regulations for at least 3 1/2 years, after hiring! We pay on a

    $832 (Avg Bid)
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    10 bids

    ...initiate a transfer via DTMF tones. The current PBX (main PBX) doesn't support tranfser via DTMF. This is the tricky part: I *think* we need to put asterisk in between the main PBX and the phone. I *think* Asterisk should be able to listen for DTMF tones and initiate a transfer to the main PBX if the correct DTMF sequence is entered on the phone. I need

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    I need a C language programmer to customize Asterisk Open Source to adapt to an owned security solution by driving communications flow from a GSM gateway.

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    Hello, We have multiple asterisk boxes that need maintenance, often passwords reset, names changed, etc. This is an on going contract to support our customers.

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    I am looking for an expert programmer in asterisk that can set all call rejection codes to 503 and Instantly reject all busy circuit calls without any delay by removing the played busy circuit massage, these mods needs to be done on 2 Freepbx Server running Asterisk 1.8.17.0

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    i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done

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    ...want to enhance the meme, including a background image (see attached samples) Choose from one of the options below to create your meme for this contest (do not include asterisk in design): *FACT: Social media platforms make money from YOUR usage through ad sales. FACT: You can legally SHARE in this revenue! Comment "more information" to learn

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    Guaranteed
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    26 entries

    Zoho's phonebridge plugin supports up to Asterisk 1.4 [login to view URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially

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    I have installed my amazon ubuntu server and asterisk server . Asterisk server is working well . I am looking for somone that can install and connect SIP so that I can send sms code perfectly

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    ...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck

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    I have all details .i need to configure sip and configure asterisk server . VIP experts

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    We are looking for an expert in Globus toolkit 6. Configuring asterisk, postfix, red media server and QGIS in Ubuntu server.

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    A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign

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    outbound calls and select options automati, will send more info. about project .

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    hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system

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    hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly

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    Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones

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    ...domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not forwarded to the same registration server. I have a budget of £100 for this project. If your config works

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    I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]

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    We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.

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    Hello I need someone to help me install whitelist on my asterisk white will reject all the calls with 503 and pass only the numbers in the list

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    Need to configure 10 internal accounts on IP phones, connect 3 external telephone lines. Set the voice greeting and the voice menu according to the technical task. Implement a CDR report and record conversations.

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    I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a

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    ...working in APK format Full source code Simple manual for compiling and generating the application from source A SIP client running on another phone that connects to the server via WIFI and able to demonstrate the functionality in this project. Features - Route call from SIP to GSM - Route call from GSM to SIP (preconfigured SIP client)

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    Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !

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    Can you make android and iOS app for Asterisk client? Similar as Zoiper for iPhone.

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    We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards

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    Need a2billing expert with proven record to do a fresh complete installation of a2billing (asterisk realtime, freePBX) and API ([login to view URL])

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    Create and train italian acoustic model based on CMU sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be con...sphinx using a set of 23 given word and audio files pronounced by 9 different speaker. Model should be configured for telephone (8 khz), in order to integrate it in Asterisk with mrcp server.

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    ...want to build a customized SIP/VoIP Softphone with one new Skin design as per choice and my logo for unlimited user license that runs on PCs. Windows. The server side it's done and use Asterisk server with PJSIP. I would like that the developer use other solutions to do that, doesn't start by the zero. I hope at least these features in the 1st step of

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    ...someone to handle the very few tech support calls I may get while out of town. Might go the whole week with none sometimes. You need to be very knowledgeable of Freepbx and asterisk in order to support my clients. I will on occasion need projects handled such as repartitioning, adding a logical volume etc... Trust is a key because I will be giving you

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    Hi, I need an asterisk guru to help me acheive my goal. When someone reach my voicemail, I would like my to get a notification on my pager. That mean, system would have to dial a number, wait a couple sec, and dial 123 then hang up. This need to happen everytime someone leave me a voicemail.

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    Hi, I am looking for Asterisk PBX expert. The person should have through knowledge of how to install configure and maintain Asterisk on linux server. We have 2 projects for which we are looking for an expert. You will be working with me directly to estimate the project and after implementation you will handover the project and explain entire project

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    This absolutely takes a programmer. After doing all the updates needed for WebRTC and changing OS on Server...OS on Servers, not much is working any more. So the system has to be brought back from near death. This is a system with WebRTC and 3 physically separated servers, Apache, Asterisk, MySQL, plus a commercial ZBF router. Nothing for beginners !

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    ...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud

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    Hello, I have installed the SuiteCRM and Asterisk system, I need to integrate these systems (call recording, connection identification, history of connections, etc.). How much will it cost a script and configuration?

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    we did integration between zoho crm with yeastar pbx "u100", and it was working successfully, but if we received any calls through our queue "sales queue or technical queue" it is not displayed calling id name, it displayed pbx trunk number so we would to make a custimze code which enable us to knows who is calling.

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