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    2,000 xmpp asterisk jobs found, pricing in USD

    Hi, I am looking for a software developer who can build real-time speech to speech language translation software. Need this done using Asterisk or FREEPBX

    $1583 (Avg Bid)
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    19 bids

    I need a graphical interface for the asterisk to do the following: Using raspbx, sql, php We use 10+ 3g modems (with the possibility to see their status and edit the imei from the graphical interface) We have a database with 6,000+ phone numbers ( imported in sql) We have a set of questions that we want to ask our clients automatically. An ivr-dialplan must be created, all audio messages must be TTS (text to speach and retrieved from the graphical interface, we must have the option to change questions over time.) We need to see in real time in the graphical interface the calls that are in progress, statistics, the date when it was called and the keys pressed. Appeals to the number xxx -> 1) xxx answers, we save the action in sql, we return the message: Hello, my name is x...

    $1091 (Avg Bid)
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    I need some help on setting up XMPP Ejabberd on a Linux system with MySQL. I already setup the Ejabberd with default database but need to use MySQL and need some assistance regarding the Ejabberd API and other configuration. We can discuss any details over chat.

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    I need some help on setting up XMPP Ejabberd on a linux system with MySQL. We can discuss any details over chat.

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    Looking for intelligent, engineer minded, polite, honest, diligent full stack developer (team). Proper English is essential. Must: all php framework, database, ftp, email, push, crone Advantage: unix server, upgrade, ssh, htaccess, centos, cwp, admin, mongo, asterisk, webrtc experience, python, rss Continuously work, bid not important. Auto-bid will be auto-refused. Please send a proper intro.

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    We have a log file where we have a number of ID numbers for the shell script to retrieve the time in the log as below cat /var/log/asterisk/full | grep C-0000335e | grep 'Spawn extension' | grep default [2019-11-28 21:52:48] VERBOSE[10832][C-0000335e] pbx.c: Spawn extension (default, Queue-206611, 2) exited non-zero on 'SIP/fcplatform-00004e64' cat /var/log/asterisk/full | grep C-0000335e | grep queue_time= [2019-11-28 21:52:43] VERBOSE[10832][C-0000335e] res_agi.c: : Query is UPDATE queue_calls set queue_time='13',agent='l/203@context-out/n',status='0',call_time=Now() where unique_id='1574974337.190819' and queue_name='206611' The script should then state where each ID number we have in a file, how lon...

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    Software to make Templates of all kinds and Invoices with Magnus/Free Side Billing. Templates are made with LibreOffice. OS is Ubuntu. You must use Teamviewer to access our server and knowledge of LibreOffice, Linux and Asterisk is a must ! -- Will be divided into 3 milestones. Everything is documented.

    $442 (Avg Bid)
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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 Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document and see attached summary for what is done so far

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    I'm looking for a XMPP ejabberd expert to customize few things . place your bid for more details .

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    i have server cloud and IP telephone i need install Asterisk free pbx to connect our getaway

    $128 (Avg Bid)
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    A php script to manually add a phone number and Queue ID into FreePBX's Queue call back option to be called back at front of Queue. Run as a pipe from bash. The idea is for the FreePBX callback module to do all the work we just need to add the phone/queue number into their system. FreePBX Asterisk 16.6.2

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    I need help for asterisk goip long term , I have special dial plans that help to avoid sim blocking , I need to fine tune these dial plans .My budget is 50$ .

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    we have an asterisk module as a POC we need to create it for production enviorment.

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    We are looking for a VoIP systems engineer to help complete the development and testing of an internally developed web based calling application linked to Asterisk/Free PBX. - Experience with Asterisk and/or FreePBX - Experience with Apache, CentOS, DNS, hosted services, MySQL - Network design – Working with Firewalls, DNS, Load Balancers - Experience in software as service architectures (SaaS) - General telephony understanding - Understanding of VOIP platforms like Trixbox, Elastix, Freeswitch or FusionPBX - Configuring various VOIP Phones and iOS/Android Smart Phones - Knowledgeable in IP Telephony, unified communications, data networking, telecommunications, video technologies and Call Center - Experience working ...

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    I need to fix a bug on yeastar tg1600 on the sms to email side. can you help me? is an asterisk appliance that stopped to send email when received sms 3 months ago I don't know why

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    This is (potentially) very easy money if you are a seasoned XMPP veteran with lots of hands on experience with the eJabberD XMPP server. I just need a step by step instruction on how to add a secondary eJabberD server as an additional cluster node to an existing eJabberD installation. You will need to answer a couple of basic "vetting" questions before I award the project as time is money and we all don't want to waste any. So please don't bid on this if you are not a real expert in this area and have done this several times already. To make sure you at least have read this project description your bid has to begin with "Yes, I know XMPP!" otherwise we both know that you skipped reading this all through the end. OBVIOUSLY I would expec...

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    I am looking for a developer who has good experience on EJabberd and android .( Customization work )

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    Hi, I have a new server that has been apparently - partially-- built. The new server is to replace a hacked / rooted server. While attempting to terminate a call through the new server the call terminator is getting error code 503. Unfortunately the developer is missing not responding.....

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    You must have at 2 years production experience in XMPP based projects. Maximum project budget is US $300. Develop chatroom web app based on XMPP server, NodeJS, MongoDB backend. Project is described in terms of UI but frontend UI development will be done by another developer. This gives what the backend supposed to provide so that the frontend developer can do his project: When a user opens the website where the app is hosted: the page simply shows login screen with just one input box where the users enters alias if the chatroom does not already have user by this alias, chatroom screen opens otherwise the user is shown the error message that the alias is already taken prompting the user to enter different alias. the chatroom UI: The main window is showing group chat c...

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    I've done a lot with Asterisk and configured many Cisco phones to work with it but this one has me stumped. I'm looking for someone that has successfully configured one of these models to work on an Asterisk server. I'm not using any GUI like FreePBX. I have a custom-compiled Asterisk running on Ubuntu 18. Basically I'm looking for remote support to get this done. Thanks!

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    i need to integrate asterisk with php to make popup window when receive new call

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    i would need some statistics from our asterisk/freepbx server we need to read from AMI directly (with node.js server) i would need a statistics cronjob which run every minute and enters data gathered into a mysql table data that shall be fetched: calls in the queues (customers waiting) calls in progress (customers calling with agents) waiting time per caller in the queue Talking time per caller in the queue we have more queues so queue info must also be stored We have already installed 1)Node.js 2)pm2 already there runs a process which processes other information so the scheme is already setup- we just need an experts who extends the functionalty

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    Software to easily make Invoices with Open Source Magnus billing and/or FreeSide Billing, for Asterisk PBX. To be used with Ubuntu, LibreOffice. Details to be discussed.

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    vtiger crm installation, customerization and CTI asterisk integration in house by an expert having minimum experience of 5 years in thi s field

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    I need OAM software and initial Topex MultiAccess GSM Gateway installation and configuration. Following steps are requied: - instruction how to connect GSM Gateway with PC - installing OAM software (software must be provided, I don't have) - network configuration - setup SIP Trunk to local Asterisk server - configure outbound and inbound routing - testing

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    Hi everyone. I need a virtual landline phone number where people can call and speak to each other 1 on 1. In the attachment you will see a little example what we are planning. Thanks in advance. Feri

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    I'm looking for a developer who has experience on XMPP, ejabbred and android xmpp client . Please bid if your familiar with above requirement

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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 Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document

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    Create VOIP server on aws using Asterisk frameworks

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    Hi, I need an Ejabberd developer to design and code server side component of a chat application. Should be familiar with XMPP, android studio and scaling of servers. Must have done at least 1 ejabberd project.

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    A very similar project sample that used an ejabberd or xmpp with SIP capabilities After the sample then we are good to award you Thanks

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    Hi, I need an Ejabberd developer to design and code server side component of a chat application. Should be familiar with XMPP, android studio and scaling of servers. Must have done at least 1 ejabberd project.

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    we'd like to build Asterisk click to call API. Please check the attached PDF and let us know if you are able to deliver the task. Note: we need ASAP.

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    I am looking to develop a click to call API to be used with Asterisk. You must have good experience with Asterisk/AMI/ARI. All details attached in PDF. Please don't bid if you have never done this before.

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    We are looking for an asterisk expert, can help with a minor thing via anydesk. We must have transferred some data to our database with each call

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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 Centos 7 *kvm ubuntu server LTS 18.04.3 or 19 *Virtual machine ubuntu 19 In the attached document

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    I need somebody to review an existing freePBX / Asterisk installation. I want to connect two Trunks properly and use two Cisco Phones (CP8861 3PCC and CP8821) with it. One Trunk and one number is set up correct and works with Zoiper on mac. All other numbers (4) are set directly to on Cisco CP-8861 (working fine), but i want to move them to my freePBX Server (hosted on a vserver).

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    ...support the same voice-over-IP protocol, and at least one common audio codec. Many service providers use the Session Initiation Protocol (SIP) standardized by the Internet Engineering Task Force (IETF). Skype, a popular service, uses proprietary protocols, and Google Talk leverages the Extensible Messaging and Presence Protocol (XMPP). Some softphones also support the Inter-Asterisk eXchange protocol (IAX), a protocol supported by the open-source software application Asterisk." Basic Fetures "A typical softphone has all standard telephony features (DND, Mute, DTMF, Flash, Hold, Transfer, call history, call outcome/disposition etc.) and often additional features typical for online messaging, such as user presence indication, video, wide-band audio. S...

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    Featured Urgent
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    I am currently sending SMS with AT commands in Text mode. In order to better support concatenated SMS, line breaks, and encodings, I want to switch to PDU mode. I need a PHP function that will generate PDU to be sent by Asterisk. Output need to be a JSON array with the PDU generated and part number, and total number of parts. Maxumum parts is 10. If more than 10 parts, error needs to be returned. If a mandatory parameter is missing or if parameter has wrong format, error needs to be returned. Parameters sent: *Destination number in international format (ie: 14152470402) *Message – SMS content Validity Period - in hours (if empty, Default is 72 hours) Status Report Request - 0 or 1 (if empty, Default is 1) Response expected: Status - ok (success) / ok (error with code) Numb...

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    I have a vicidial system up and running but yesterday as we making calls we stop getting sound and calls were dropping. Error message we getting on asterisk is SRTCP UNPROTECTED FAILURE

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    I have a vicidial system up and running but yesterday as we making calls we stop getting sound and calls were dropping. Error message we getting on asterisk is SRTCP UNPROTECTED FAILURE

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    I have a vicidial system up and running but yesterday as we making calls we stop getting sound and calls were dropping. Error message we getting on asterisk is SRTCP UNPROTECTED FAILURE

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    Needs integration asterisk server with odus. Ai

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    Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future:...

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    need a long terms sysadmin expert in: SYSADMIN Linux: -freeipa -freeradius -general linux debug -ODOO debugger -php installation knowledge -apache installation knowledge -CentOS expert -docker expert -asterisk expert -freepbx expert -routing expert ( we use mikrotik routerOS ) -apache / mysql / postgre SQL -Google Cloud Platform expert for managing application, instance, docker, db etc... SYSADMIN Windows: -AD expert -various problem solving like antivirus, printer etc... if not have full knowledge please dont bind an offer...

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    We would like to use the API provided by Mobile carrier to do the following... 1) Activate SIM Cards for Mobile Services directly from our service portal We will need this API incorporated in our A2Billing Server using agent portals you will need to know Asterisk programming and know about A2Billing Software. We are ONLY looking for developers who have done this type of API development! We will provide server and access needed for this project. We will provide detail documentations for API. We will provide Details Document of the API once we accept bid. or speak with you. We are NOT looking for any website to be develop, so please do not offer.

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    About us: Negombo Park has an Italian website. We're looking for an experienced translator to collect all the website text and translate it from ital...link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project doesn't include installation but localization and collection of text will follow two distinct milestone, where we'll perform a check for any missed text (for example, popup di...

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    About us: Negombo Park has an Italian website. We're looking for an experienced translator to collect all the website text and translate it from ital...link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project doesn't include installation but localization and collection of text will follow two distinct milestone, where we'll perform a check for any missed text (for example, popup di...

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