Asterisk elastixlavori

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:...

Hello, We need help to setup Analog Voip Gateway with Asterisk, model: ~ Remote session is provided with TMATE, so be ready for it. Thanks in advance

We are looking for an asterisk developer to help scope and implement a project

Due to hardware problems our very old Asterisk server died on us about 9 months ago, but due circumstances we haven't been able to replace it. We are a small company with just a few phones and some other linked equipment so we made do using a direct line, our mobiles and a professional skype account. However we would like to get the situation fixed before the end of the year if possible. Due to the situation the Asterisk will need to be set up on a fairly new server (2x XEON E5-2620v3 6 Core 2.4GHz/32GB(2x16GB)) but with some old(er) peripherals like a DrayTec Vigor 3300V switch which holds the FXO and FXS connections, some grandstream SIP phones and a few regular PSTN phones, as well as Softphones on our laptops and Android phones. Also there is a SIP door intercom we ...

Summary: Our current application is housed in Wordpress. The main applications backend is connected to an Asterisk system. We would like to switch over to a Bootstrap framework and update the template used for our website. Deliverables: - Transfer the Asterisk-based application that's currently housed in WordPress to a Bootstrap administrative template (we'll provide the template with the backend admin layout) with the same read & write capabilities that are housed in our current Wordpress setup. This includes: Read & Write access to subscriber information, ability to change plans for subscribers, etc. - Update the current website to a new bootstrap template. Seeking an experienced developer, that has experience with Bootstrap, Wordpress, HTML, CSS, &a...

Software to make Templates of all kinds for Invoices, etc. To be used with Magnus/Free Side Billing. Templates are made with LibreOffice on Ubuntu OS. PBX is Asterisk 16.6. Work is to be done with Teamviewer on our servers and good knowledge of the new LibreOffice, Linux and Asterisk is a must ! -- Everything is documented and the project will be divided into 3 milestones.

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

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...

I have a SIP Trunk for Saudi Telecom Company (STC) in Saudi Arabia and I want to configure inbound and outbound calls. The SIP trunk comes with static IP address and its connected directly to the STC company. I need an expert who did this STC SIP trunk connection many times. I need him/her to send me the full configuration steps so I can do it. Note: 1- I can Not let you access the Elastix server so Please do not ask for that. 2- The line must detect when the user answers the calls and when the user hang up

Diseñar portal web sobre asterisk que incluya opción registrarse, login (ingresar), DID para seleccionar o escoger (activar).

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.

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...

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.

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

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

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

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$ .

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

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 w...

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

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.....

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!

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

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

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.

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

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

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

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

NEED SOMEONE TO MAKE ME A WEBSITE FROM SCRATCH ELASTIX WEBSITE

Create VOIP server on aws using Asterisk frameworks

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.

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.

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

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

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).

...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. Softphones provide a variety of audio c...

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...

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

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

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

Needs integration asterisk server with odus. Ai

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:...

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...

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.

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...

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...

FreePBX (Asterisk) Freelancer in bangalore, India

...need a sip-bridge application which will work in android. 1. Calls will start from asterisk with sip protocol. 2. The sip-bridge application which will work in android will take the coming sip requests and make the calls via skype/viber/bip. Calls will be transfer to the mobile number, not his skype/viber/bip (to the called person's mobile number). The info about which number is gonna be called will be in sip request which starts from asterisk. person who originates the call via asterisk will make a phone call with the person being called via his mobile number with the help of sip-bridge application and skype/viber/bip that sip-bridge application; all the sip signalisation messages between asterisk and skype/viber/bip should be exactly transfer end to...

ive issabel asterisk 11 and vtiger 7 installed and configured i've even configure Vtiger connector.. nowi just need you to configure the dialplan so the users will start getting the pop on on vtiger Inbound +Outbound (Click 2 Dial or Click 2 Call) these 2 should work