Asterisk PBX jobs
I would like to configure my asterisk server and GoIP gateway, make call from SIP client through gateway with caller ID which is defined myself
I have some work, in an Excel spreadsheet. Type a combination of numbers and calculation operators, like the plus sign (+) for addition, the minus sign (-) for subtraction, the asterisk (*) for multiplication, or the forward slash (/) for division.
I have some work, in an Excel spreadsheet. Type a combination of numbers and calculation operators, like the plus sign (+) for addition, the minus sign (-) for subtraction, the asterisk (*) for multiplication, or the forward slash (/) for division.
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2...
I am in possession a phone cisco cp-9971 and I need to configure it to run on an asterisk server. In order of importance, I have to configure outgoing calls, incoming calls and video calls.
We would like to invite bids from prospective developers to design a website and online store where we will sell telephone numbers, plans and the website will capture customer details, interface with a payment gateway, interface with a fraud/verification gateway, keep call logs and messages, interface with a pbx system via a script to make purchased numbers active instantly. we would like the new website to be a hybrid between and The customer buys a number as it is on and activates / connects it instantly to an alternative number keyed in via the web platform as it is obtainable on voicenation.com. Our database gets updated instantly.
Configure Astlinux Project , for call forward to external number , IVR , message .
I have am existing FreePBX server that was using Flowroute Trunk/SIP/Route. I switched to Twilio as provider. Need help understanding config for lines. Please setup 1 line working fully and document steps to provide, I will then test and do the rest. More PBX work to the person who wins this bid, if wanted.
rebuild an asterisk server with AGI. all configuration and files are ready and OK. just configure the AGI execution
I am looking to for a modern website which can accommodate my ordering platform (WHMCS), and have it geared towards my industry - hosted PBX and business services (web hosting, email, hosted VOIP solutions, and internet). Our existing website is a template of ThemeForest - we're at a point where we need an original design to differentiate ourselves from the competition.
Modify configuration file for existing production router to allow NEW VOIP provider's WAN traffic to pass through router to PBX equipment on LAN. Existing config and ports required to be added to configuration will be supplied. Freelancer should make changes and test in their own environment, then send finalized config with ADDITIONS AND CHANGES clearly marked for use by onsite IT staff.
I am having an issue with the ivr on fusion pbx. Some of the prompts work great, and as soon as i hit a digit to go to another prompt it just hangs up on the call.
The project is about integration of asterisk with Cepstral text to speech engine.
I need an application build on java using asterisk at backed for making and receiving calls. A web based phone dialer and notification on web page for incoming calls is required. Initially I want this then later in phase 2 we will integrate it with a complete call center application.
hello, i want to bye a custom asterisk dialer that provide this fonctionnality: get list of users from database call users on order log call status (success,failed,answered, action taken) press key send set number of simultaneous calls make a miss call with wait time (8seconds) please response urgently
We have a CentOS file server that we are looking to use for storing call recordings from 2 smaller CentOS FreePBX/Asterisk machines. The files are currently stored in /var/spool/asterisk/monitor on the 2 local machines. We'd like the files and directories currently in that directory on each machine copied to the file server into /root/customername/. We'd then like NFS setup on the file server and 2 PBX machines, so that the /var/spool/asterisk/monitor/ directory and its sub-directories on the PBX machines is mounted from /root/customername/ on the file server. The configuration needs to persist upon reboot.
I need a professional expert on Asterisk 14 to develop Callcenter module. We will make the theme template available. The code should be fully commented and documented.
We need to build an asterisk connector using AsterNet as a dot net application that can communicate with asterisk over AIM
Hi, I need full hands on asterisk training including security protection etc. I will also need training on Mysql database along the side. Thanks
Create Application or Web based app get list of users from database call users on order log call status (success,failed,answered, action taken) press key send http request set number of simultaneous calls
Create an Elastix (version 5.0) IP PBX on Amazon ( EC2). Installation should be done using latest CentOS 5.5 (or newer). You will need to provide logins and IP for the GUI. Alos need to receive the Amazon AMI so that we can create several Test Boxes (PBXs) Payment of the milestone will be made after the testing of basic functionality. You will need to have experience with Asterisk ( Elastix), Linux and amazon cloud so the entire process should be delivered in a few hours.
...geocentrically in an Android and iOS app that takes a picture and process OCR of a NationalID with grided photo while aiming the camera. the OCR process could be in the device or in the cloud. This app connected to a Drupal system for admin and display the data, several hierarchies are needed. also the ability to enable a dialer module that connects Zoiper or another dialer that runs with Freepbx or Asterisk from Drupal that dials a cellphone number stored in the database when registers an ID it asks a dialogue in the app that asks for a 10 number cell phone number. also perform SMS messaging to an ID or a group of IDS. Data stored like street, city, state will be processed using MapBox or other tool for displaying a pinpoint map, select and process dial or SMS messaging. all dru...
raCreate Application or Web based app get list of users from database call users on order log call status (success,failed,answered, action taken) press key send http request set number of simultaneous calls
I have IP - PBX with sip trunks and sip client. We need an expert in voip implementation. Hardware we use for FXO are: SPA-3102 HT882R Old panasonic analog pbx. There are few issues. 1. We had a problem with elastx we have had up and running for 3 years and decided to reinstall. After the installation, incoming calls can reach sip client but outgoing does not happen 2. We use chiffon sip client on android and it does not give you ringing sound when you are waiting the other party to answer. 3. We used SPA-3102 with the analogue Panasonic pbx intercom but now cannot get intercom to ring extension. 4. Cannot figure out Panasonic PBX setup for number of ring for intercom 5. Certain international outbound to go though particular trunk (2talk) 6. NZ M...
Hi, I have a huge issue with my asterisk server would you be available to discuss this today? Thanks Rene 8 0 1 6 0 8 8 8 6 3
Hello There, We are looking from someone who can integrate Viber call s to Asterisk, Were we need any one who calls us on our Viber number to be connected to our Asterisk. Very Simple you are free to choose the way you want to implement this task, like sip trunk or any other tool you may need
It is necessary for 7 phone providers' sip trunks to be directly registered on kamailio and the asterisks must be allowed to make outbound calls selecting one of the trunks
i want to update my ubuntu package useing command apt-get update its show error main package 404 not found i just want to update package and install asterisk & openvpn if you can help to to fixed package issu please bid
Seeking to have an API developed to be able to read and write presence data into Office 365 Skype for business. Eg when user is on call like to set external pbx busy lamp to busy & reverse presence also required as in if on external pbx call require to set Lync/Skype for business presence as busy. Project needs to support Office 365 version of Skype for business as well as on premises version.
i have install freepbx on rasqberry . i have a sip number from a provider and i want install my sip settings on freepbx to do and receive calls on my 3 local extensions . my local extensions is ready , i need only sip configuration for inbound and outbound route .
I have one server of Elastix and i want to enter 20 IP that i have one list, and 4 outbound IP. All ports have close and the server have to secure.
Currently have a working FreePBX install on Sangoma hardware. The hardware is old and not rack mounted. I have new rackmount hardware (online, ready, with IP address, free PBX installed etc) the following needs to happen to it; Port the current working config to the new hardware. Re-configure the phones to point to the new hardware. The phones are Yealink T28P phones On the handsets, make the BLF fields light up as if their line keys (Line1 Line2 etc - 1 per SIP trunk) - i have 5 phones, if you can make one work i'm sure I can copy the config. Configure the GSM card (Sangoma W400) - The new hardware has a 2 port GSM module in it, which will be used for voice and sms, so configure the SMTP reciever and the voice sims as a trunk etc. Configure the 2 analogue trunks & 2 anal...
I would like to order a setup of Zabbix server which will be dedicated for automatic PBX monitoring within a separate VLAN. Measured values and triggered actions below: 1) Disks, PBX operation threats for HDD overload. 2) ETH analysis, if there is a bigger load of interface than the declared size – start a fraud alarm; 3) RAM analysis, PBX operation threats for RAM overusage. 4) Simultaneous connection analysis, threats of PBX operation under the scope of overload of simultaneous connections (filling the purchased bandwidth in Zabbxi, alarm at 75%); 5) Is the phone available, meaning whether the device is available wthin network, continous statistics of device operation, whether the DND option is enabled, forwarding, if DND enabled – is it possible ...
We have Asterisk server which can handle WebSocket protocols. Our task is to connect JSSIP to the ready design and layout for the work of the site. Documentation reference Http:// Links to the library: Https:// Demo: Objectives of the project: -User login with defined credentials. -Providing a point-to-point call through the browser. -Providing chat. -Presence function. -WebRTC Video Quality configuration (with predefined qualities and video camera and microphone switching) . - Indication of ringing and busy state. -The integration code must be copyright, not stolen. Layout: -crossbrowser, cross platform adaptive layout -adaptation for mobile devices (recommended bootstrap layout
Create a Program that would help a student pick the classes throughout college until he/sh graduates prerequisites must me taken into consideration i.e if cosk1020 is a prerec for math2070 math2070 cant be completed until cosk1020 is on check sheet one asterisk is prerequisites 2 asterisks is co-requisites
Create Application or Web based app get list of users from database call users on order log call status (success,failed,answered, action taken) press key send http request set number of simultaneous calls
Minor changes including: - Adding a recording if the user doesnt dial the right number on their phone. - Letting the user try again. -If no # dialed on the phone and hung up return a 0 to the php based dasboard.
please find below project requirement which we need to add in the fusion pbx 4.0 running on free switch.1. Project Scope:Enable API setting to send inbound caller ID calling party information for any domain/tenant as fully supported solution.2. Use case:Client want to send a caller id information through API to Icabbi for domain configured inside fusion pbx tenant domain name olympiacentrewest.local3. Example scenario:Calling party with CLI 075072378510 calls inbound route for DID 02073867810 Call will ring any extension/ring group/queue/external DID pbx will also send calling party number through API as shown in below link with international Enum format 447507237810 and call extension number example URL is as follow
I have "Asterisk 13.1.0~dfsg-1.1ubuntu4" its been working fine as my private voip server. However I would like to use the sms / messaging feature. I do not have any GUI or 3rd party addons this is a stock Ubuntu Asterisk install. I am using Zoiper on Android and would like to use the SMS / messaging feature. Also my DID supports incoming/outgoing SMS so would like that to work. I would like if possible some one to guild me through what I need to do to make this happen. I cant give access to the server but can give the files u request (less any private info in them) Please dont just point me to some readme u googled as I have tried a few with out luck. So this will most likely need some one with experience.
I have a voice blast software with code written in asterisk and Php. It is integrated with an application and on hangup of the call, i am sending a hangup message to the application. The code functionality is working fine except in one scenario. While the call is been executing, if the customer disconnects and it moves to GoSub return context , in this particular scenario rest of the dial plans are not executing. It will hang up the local channel log for SIP/Dahdi channel hangup so i am not able to trigger the hangup message to the application. It is an urgent requirement to be started today. Only bid if you are an expert on asterisk dial plan.
please find below project requirement which we need to add in the fusion pbx 4.0 running on free switch.1. Project Scope:Enable API setting to send inbound caller ID calling party information for any domain/tenant as fully supported solution.2. Use case:Client want to send a caller id information through API to Icabbi for domain configured inside fusion pbx tenant domain name olympiacentrewest.local3. Example scenario:Calling party with CLI 075072378510 calls inbound route for DID 02073867810 Call will ring any extension/ring group/queue/external DID pbx will also send calling party number through API as shown in below link with international Enum format 447507237810 and call extension number example URL is as follow
Hi, We have a Live Asterisk panel that need bug fixing.
ive got freepbx on a server and am using paid version of vtiger. need to configure the asterisk credentials in vtiger. 20 minutes of work.
