Iax asteriskprojekty

we have asterisk 11(freepbx) setup and running. need vtiger install and connect to asterisk.

We are in a requirement of sending SMSBlast/Campaign by using the SIP account. We have VICIDIAL and ASTERISK EPBX. So, by using any of these SIP accounts need to send SMS without gateway.

This project is to setup a FlowRoute SIP Trunk for inbound/outbound calling and one user (for a softphone) on a freshly installed Asterisk based server (XiVO).

We want to 3D print a custom designed stair/step on our SPIRAL staircase. - The primary focus should j...designs in a .STEP file - bonus: apply generative design / topology optimization study principles to the design for an organic and unusual look. Here are some samples:

Expert needed to install and configure Goautodial Call center software. the platform is based on linux centos 7. Used technologies : WebRTC Asterisk Voip

Hi, I have a asterisk PBX and my "programmer" really just did a very poor job. Basically, we moved the system from 1 server to another and again when he transfered it he did a very bad job of it. 1. The system needs a2billing added (it was on the previous one) 2. The system needs to be updated tp php7 (previous version on old server was php 5) 3. The system needs all the - customized scripts-- to work with php 7 on the new server The domain name needs to pint to a specific index file (in a folder on the server) The same file the IP address points to. He renamed a few folders also (where both the previous folder and his -- renamed edited-- ) versions are on the site. This way you can see what it - should have been-- and what he has done that does not...

Installation and customization of Asterisk 1) Install asterisk on server 2) Customize Asterisk to work as per below logic --> Asterisk should be able to receive inbound calls on DID number for different providers --> When there is inbound call on asterisk , I want system to hang up inbound call and automatically dial out/Call back on caller id from which inbound call was received. --> Once asterisk dial out then it should be capable of identifying whether call was answer by Answering machine if it is answered by answering machine then system should hang up call without forwarding to agent extension , Only calls that are answer by Humans should be connected on agent extension. In simple words i need Asterisk with Callback Featu...

We need to configure a Cisco VG224 to work in ISSABEL with all 24 analog ports. Configure dial plans and to work with more than one Cisoco VG224 in the same ISSABEL

Hi, I have a working freepbx installation. That server is working since two years without any issue. The version of the installation is 14.0.13.12. Now I want to use same freepbx server for message communication. I use X-Lite as VoIP client. I want to use X-Lite for message and file transfer as well. All extensions connected to the server could communicate each other via messages using X-Lite. Regards, ALSOFT

Need an experienced tech for Vicidial support in real time provided over Skype and email. Must be available at any time 9am-5pm PST and respond within 5min during these hours. Work load is minimal and most days does not take more than 10-30 minutes combined but must be available during these hours without exception. Must be experienced with Vicidial. Must have good conversational English. Please offer monthly price.

I need A2Billing/Freepbx Software installed on my server and configured for a soft-phone, which is a third party app on my server, the objective is that I want to:- 1. I just need help setting up dial-plan to call long distance in USA/Canada and all International and Local calls. 1-a. Dial remotely into the server, 1-b. Enter a pin number, once the pin number is entered once, thereafter, set-up a caller ID authentication on subsequent calls ! 1-c. Enter the destination that I want to call 1-d. Then the soft-phone would initiate the call. 1-e. Call history 2. Configure a method to discourage Caller ID spoofing and to activate/deactivate account - prohibit Spam Robocalls 3. Enable Search feature for admin to search by email, Account Number and caller ID, Pin Number and Account Number 4. Cre...

We are MVNO project. Asterisk Deployed. Need to connect Kamalio and SS7

I need to set a signaling gateway (like kamailio/drachtio etc) that will be use as a proxy between Asterisk server and web clients. 1. Users will use webdialer based on webrtc that will connect to the proxy using username and password. The proxy will route the signaling+RTP to specific asterisk server as a standard SIP client. The webdialer will be able to initiate calls and receive calls. The webdialer will have in addition "hold/transfer/conference" buttons. 2. CallMe button that will use a subset of the webdialer. This button should be generate a token that contain the domain, username and password. This button when clicked will connect to the proxy and the proxy will dial the preconfigured number. I'm expecting to get the server installation process and c...

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

Hello We need to develop a SIP to Viber/Whatsapp gateway. The gate...of the application, no matters on the version number. The implementation should return the correct call error codes to the SIP backend, i.e. CALL SUCCESS, BUSY, UNAVAILABLE, etc. For a successful project we'll select the one, triggering successfully continuous SIP/Viber calls. Functional flow 1) Calls originating will send to Viber/whatsapp gateway 2) Viber/whatsapp gateway converts the sip/iax signal to viber protocol 3) the termination number carried from the origination header will be checked by the Viber gateway , if the number is used by Viber/whatsapp and if the number is online, the call will terminated on Viber/whatsapp!. 4) if the number is not used in VIber it sends 503 error and rerouted to ot...

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...

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 working ...

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

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 ...

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...