Asterisk dialplan jobs

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    11,439 asterisk dialplan jobs found, pricing in USD

    We need installation of newfies dialer, to get 2000+ concurrent calls from each box, it should be build on VM enviorment. Also asterisk amd module and beep detection modules need to be install.

    $50 / hr (Avg Bid)
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    3 bids

    Hi Everyone. I need your help on the following : #1. Configure SIP trunk #2 Call forward to mobile phone numbers with a strategy in Freepbx or 3cx, Asterisk or other VoIP services . A strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk.

    $87 (Avg Bid)
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    15 bids

    Need a very experienced asterisk developer for robot calls. Existing system is preferred. Using my individual asterisk server, sip accounts, multiple channels - has to be integrated to basic crm system. No twilio, no nexmo, no any 3rd party app. Any general bid ("I'm good in wordpress and shopify etc." OR " I can adjust twillio, nhx or similar") will not be considered.

    $2446 (Avg Bid)
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    19 bids
    Java application 4 days left

    Create and test an application in java that will allow a user to create a text file to store data values associated with Player objects. The application will provide a menu for a user to display an encrypted list, display a decrypted list, add, update, and delete the Player data record in the text file. The menu will also contain a help command for the user to display the menu and an exit command ...

    $87 (Avg Bid)
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    9 bids
    Trophy icon ASTERISK AMI 4 days left

    Hello, we need asterisk AMI script (syntax) for yeastar PBX we need functions must work via triggering AMI commands described and tested : 1. hangup 2. mute 3. attended transfer 4. hold please only serious freelancers with experience. Please be aware that Yeastar PBX has limited manager commands

    $260 (Avg Bid)
    Guaranteed
    $260
    1 entries

    Hi, I'm having lots of problems with FreePBX. Trying to update: Unsupported Version of Asterisk, You need at least 11.11.0 you have 11.8.1 Running Amportal: amportal a ma refreshsignatures Fetching FreePBX settings with gen_amp_conf.php.. /usr/local/sbin/amportal: line 52: export: `®': not a valid identifier /var/lib/asterisk/bin/freepbx_engine: line 100: export: `&re...

    $22 / hr (Avg Bid)
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    9 bids
    Help for Asterisk-FreePBX 3 days left
    VERIFIED

    I have an Asterisk-FreePBX System with multiple disks that needs some fixes. If You are a specialist in this field, lets talk. --- This is not for people who think all the answers are on the Internet ! This is for experienced specialists. Requirements: Asterisk, FreePBX, SSL Certificates (Letsencrypte), Apache, multiple disks on system, web dev, PHP, etc. Will divide into milestones.

    $357 (Avg Bid)
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    18 bids

    Dear Coders, We are looking for a programmer who is familiar with Voip Systems and with the following : (1) SIP: (1)(1) Setting UP SIP Server. (1)(2) Operation of SIP Server. (1)(3) SIP Express Router. (1)(4) Asterisk. (1)(5) VOCAL. (2) Protocols: (2)(1) H.323. (2)(2) SIP. (2)(3) Media Gateway Control Protocols. (2)(4) Proprietary Signalling Protocols. (2)(5) Real Time Protocol & Real Time C...

    $2848 (Avg Bid)
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    Hello, I have an asterisk PBX vers 11.22.0 . I am using a Polycom sound point IP 650. All works fine except for the transfer button. The transfer on polycom use SIP REFER to transfer the call. This is not working. Need help from anyone who know about the subject. Please respond to this project with "What up Dingo" at the beginning of your message so that I know you have read.

    $145 (Avg Bid)
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    3 bids
    speech to text 1 day left

    Hello I have Asterisk dialer and I need to set up speech to text transcription (ONLY) I use to use to use IBM watson api for this, but it has become too pricey. it is 1 Min length audio of ivr recordings each. But total millions of files. every 2-3 months 7 million 1MB, 1 Min audio files.

    $52 / hr (Avg Bid)
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    15 bids

    Hi We are looking for a freelancer experienced in Asterisk. Current developer works at another job, so you will work with me for a long term if you want. hourly rate is 25~35. 40 hours per week Thanks Anthony

    $22 / hr (Avg Bid)
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    28 bids

    This is an on-going project with various tasks managing asterisk Please apply only if you have experience in this.

    $32 / hr (Avg Bid)
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    15 bids
    FreePBX predictive dialer 1 day left
    VERIFIED

    I need someone to teach me how to complete following task with FreePBX: 1. How to make queue as a outbound call. Where to put list of customer phone numbers to be dialed by FreePBX and connect to queue. 2. Predective dial and then connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 15.0.16.18. Instruction should be based on FreePBX correct version but not ...

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    We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip. FreePBX is a virtual machine with a pub...

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    Profile description Hosted PBX Call Center solutions VOIP SIP Trunking Softphone Configuration Database

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    14 bids

    Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho VPN Both side working perfect over sonicwall...

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    Looking for Asterisk, FreePBX with WebRTC specialist to verify and fix existing PBX. This is no place for amateurs !

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    I have a running asterisk PBX - i will to rebuild new one with asterisk. i use Asterisk API, Databases.

    $278 (Avg Bid)
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    Please only bid if you have experience in asterisk with python rebuild asterisk server using backup files sip and dialplan database restore AGI (python) restore all default functionalitys Add database entry for user action

    $508 (Avg Bid)
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    migration of asterisk to clean server

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    migration and debugging asterisk

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    migration to asterisk 16 some scripts to get up and running

    $28 / hr (Avg Bid)
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    I am migrating my asterisk server to latest . I need some help to resolve the issues

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    we need an expert in Networking to create a VPN between 2 servers 1. both servers are a VM in a windows 2. Server #1 we have asterisk that need to allocate an Public ip [login to view URL] #2 is connect to local GSM Gateway to test it we need to make a call via the asterisk using the vpn and ending up at the gsm gateway

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    Resolve issue with “Cannot Conner to Asterisk” error. Update server and enhance security.

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    I need someone to help configure the Patton SmartNode 4114 (FXO Port) with Freepbx/Asterisk so we can use SmartNode 4114 as a VoIP gateway for PSTN lines.

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    5 bids

    we need to get support for any one who know the chan_dongle and asterisk good

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    I need to set a gateway that will be use as a proxy between Asterisk server and web clients. User will log to the gateway and the gateway will connect it to the specified server with SIP user and password. I'm expecting to get the server installation process and code with the client side code that provide credentials login. Once client will connect he'll be able to call and get calls usi...

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    We’ve got an asterisk system with two trunks. We need some extension configuration changes and some ongoing support

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    I want to change A2billing AGI to FastAGI due to performance and scalability. I need a very experienced person with a2billing and ofcourse asterisk.

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    5 bids

    I wrote a Script that returns a True or False boolean according to a lookup number from a website. I need someone to write an AGI on my [login to view URL] on my Asterisk server, whose main purpose is to forward a Dialed Number to my Script in Python, and afterwards if the number dialed is returned with True, then the call should be allowed to go through. If the number is returned with False, a 50...

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    hello, we use a "less secure app" with our Asterisk PBX voicemail to email message notifications using gmail. gmail wants users who use this option to make them more secure. if you google "less secure apps" and gmail / gsuite you will see what needs to be done. this is what needs to be done: The G Suite Team <gsuite-noreply@[login to view URL]> Tue, Jul 30, 12:16 AM to m...

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    Convert files from wav to mp3 files after a call is made, historic data and new data automatically after a call is made. when change is made I want to see and hear from crd reports. review and clean log data from /var/log/asterisk make rule to minimize log file size /var/log/asterisk/fail2ban /var/lib/fail2ban /var/lib/fail2ban/[login to view URL]

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    We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

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    we need an expert in call termination that have expertise in 1. DINSTAR equipment [login to view URL] VPN 3. and avoiding DPI 4 asterisk we need to set up a rout

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    I started to build this web based SIP phone using [login to view URL] - [login to view URL] - but other work has left the project incomplete. I need the project completed and updated to use the latest version of [login to view URL], 0.15.6. You will be provided with FTP access to the current source files including the HTML, CSS and current JS files. Additionally 3 SIP accounts will be made avail...

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    We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

    $378 (Avg Bid)
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    8 bids

    We need to create a custom asterisk channel, that work like chan_console, but instead to use local audio card it will use a spice connection to a kvm server for bidirectional audio. What is played from the virtual machine must be forwarded to the caller and audio coming from the caller must be streamed as microphone of the virtual machine. *Asterisk version 16 *Operating system for asterisk Linux...

    $466 (Avg Bid)
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    3 bids

    Hi Ibrahim Ali M A., I noticed your an expert in VOIP and asterisk. We are having issues with our VOIP system - in particular outgoing calls through SIP trunk are getting cut off in 6 minute 39 seconds. Asrerisk server running on CentOS. Can give access through SSH.

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    HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK...

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    HI, I need training on Asterisk and Linux i want to start voip business i need to learn command lines and also mysql and also learning about asterisk ans sip and also learning about more into Interconnects. I am looking at online cloud servers and i also require indepth training Asterisk Training & Mysql & Learning about VPN / Prioxy Servers / Integrations/ DID Numbering I perfer UK...

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    2 bids

    I’m looking to setup a Hosted PBX company and would like help from a developer who has experience performing this service.

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    To consolidate our different projects, we decided to write our own backend service to Asterisk PBX by utilizing the AGI specification (see https://wiki.asterisk.org/wiki/display/AST/AGI+Commands ). Because of performance reasons, this back-end should be implemented by using plain C, with as less external libraries as possible. We aim to use this service on a broad range of hardware, so it is imper...

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    We need to setup the integration with Asterisk 13 and Google Dialogflow to connect the Dialogflow bot to our telephony service. We have test environment with Ubuntu 16.** server with installed Asterisk 13. There are some add-on to make that integration: 1. [login to view URL] 2. [login to view URL]

    $130 (Avg Bid)
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    We need to setup the integration with Asterisk 13 and Google Dialogflow to connect the Dialogflow bot to our telephony service. We have test environment with Ubuntu 16.** server with installed Asterisk 13. There are some add-on to make that integration: 1. [login to view URL] 2. [login to view URL]

    $683 (Avg Bid)
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    - Download and install Opensips; - Create template configuration for Customer (originator) and Vendor (destination); - Customize origination peer to analyze incoming destination number and identify destination number area code; - Based on identified area code, search randomly for a number in a specific file for a number to replace at FROM field (caller ID); - Send call to destination;

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    kindly note we are call center company have UCCE and Asterisk systems and would like to integrate below systems with 1- Implement Media Server using UniMRCP 2- Integrate a TTS (text to speech) with UCCE (Unified Cisco Communication Enterprise) and Asterisk 3- Integrate a ASR (automatic speech recognition) with UCCE (Unified Cisco Communication Enterprise) and Asterisk. 4- integrate Arabic AI solu...

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    kindly note we are call center company have UCCE and Asterisk systems and would like to integrate below systems with 1- Implement Media Server using UniMRCP 2- Integrate a TTS (text to speech) with UCCE (Unified Cisco Communication Enterprise) and Asterisk 3- Integrate a ASR (automatic speech recognition) with UCCE (Unified Cisco Communication Enterprise) and Asterisk. 4- integrate Arabic AI solu...

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    Looking for a simple SIP dialer Mobile Application for iOS and Android which can register to our asterisk server and simply make and receive SIP calls. Having G729 codec enabled is preferred, otherwise GSM codec is required.

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    Need help with Vicibox. Have been getting some errors. There must be some mistake in the configuration I must have done. The calls go through fine, but when the customer disconnects, nothing is recognized in the asterisk.

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