Asterisk pbx jobs
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.
We have an extension which has a voicemail box with VMX locator on which option 1 get's forwarded to a cell phone. We need when the caller presses 1, the call should go to cell phone and the recording of the call should get emailed to a group of users.
I'm looking to develop a router capable of supporting IPSEC/OpenVPN/MultiCast to serve as a VPN between a local PBX and a remote site. The VPN and PBX will be hosted in a Xen Hypervisor network. Goal is to allow the VPN server to maintain remote VPN connections so phones and computers can communicate back to the hosted pbx without ports changing, etc.... I would like to use existing open source technology to keep deployment costs down. At the end of the day someone with great experience with VyOS, pfSense or another comparable router and VPN would be perfect.
From an Apache host create a simple PHP website that connects to a separate Asterisk software PBX and shows a real time display of the extensions that are in use and displays incoming called ID. It would need to display the Extension, user, status, incoming/outgoing, number and directory entry in a simple table format. It would also show the incoming number / directory entry for any incoming call separately. The idea being that if required this could be displayed on a large monitor so that users can at a glance see the status of each user as well as caller ID for incoming calls. There is a LDAP directory on the Asterisk system numbers need to be compared to this so that the directory entry is shown instead (or maybe as well as) the number. A demo system wo...
We need someone to show us how to provision phones with Freeswitch and Asterisk if possible. We have a running in production FreeSWITCH server, just need to work on getting it setup. Same goes for FreePBX servers as well. Phones are mostly Cisco but we can work with any brand based on recommendation. Additionally if you have the skillset we need you to integrate kamalio in our current production system or into a new environment where we can move our customers going forward.
Working Chan_dongle box I can see incoming sms and ussd in cli but they do not come in log file. I think it is some mistake in
hey, I have an old version of an opensips server not use, its just there incase i need it, its infront 2 asterisk server was using for LB and sip registration, wanted to update it to the new ver of opensip can you do it?
Need to be able to live transfer call with originating caller id. Part 1. Agent will call Customer, and tell customer we can help with service and will transfer to Specialist. Part 2. Agent dials specialist while customer still on line. Specialist answers, we make introductions between customer and specialist, and disconnect from call. Allowing Customer and Specialist to continue conversation. Part 3. Very Important: When Agent is dialing specialist, Caller ID must show Customer Caller ID. Part 4. Very Important: Must be able to have access to entire call recording, from initial call from Agent to Customer, Agent to Specialist, then Specialist and Customer full conversation even after Agent hangs up. Thanks! Happy Bidding Rob
We need to configure one server which can accept voip call and terminate on a CENTOS with Asterisk 11 sip tls . authentication is done on cli basis
We have Asterisk 13 using FreePBX 13 and VTiger 6.5 source I can make outbound calls from vTiger but I can't get the inbound calls popup showing on vTiger for users. Also I need the PBXmanager populated with the CDR details only for user of vTiger not populate for all calls into Asterisk Have TEST system ready with Asterisk, vTiger and FreePBX for you to make the changes. Require all source code and instructions on what changes were made so I can make the same changes to our live system
hello guys. who can install us asterisk, german admin interface and visal pbx manager that a beginner can handle everything? its important that asterisk is secure and not be hacked in 2 minutes :)
...configure it so the Asterisk reload over night. Also nee to install Voice Recognition (CMU Sphinx) for IVR. Dialplan for IVR will provide the option for Customer based on some criteria than send the caller to custom destination which will work with the voice recognition to ring the requested group. Allow user till 3 attempt, if it fails than it will ask user (announce) to enter 3digit service code or press defined character to rout the call to an agent. - I will set the service code for the user and display those in a directory. Check the attached document before you submit proposal. Please be specific or your bid will be rejected. There will be more task if you can accomplish this primary task. **** Submit proposal if you have hands on experience in FreePBX, Asteris...
Phone has 8.5.4 SIP firmware already. Need XML to enable features for BLF, parking, speed dial and NAT. Need TFTP provisioning instructions. SIP Server is a commerical asterisk implementation.
