Asterisk pbx server jobs

Hello, We have an issue with our VoIP (Asterisk / PBX) based application that it receives no calls when running on the background. The issue is currently that; 1) APP is in the Android recent-tabs it sometimes receives calls / broadcasts. 2) APP when not (closed) from the recent-tabs it never receives calls / broadcasts. 3) APP crashes when not directly

Hi Vladimir U., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I need to move our Asterisk system to FreePBX on Microsoft Azure cloud. I'd prefer to install it on Centos 7. I also will need to create couple extensions and connect them to a test account. There is also some basic ACD function: Dial

anyone who has worked with asterisk or a2 billing before please bid and i will give details of project

I need you to configure asterisk to receive calls and play an audio. Also to make calls. I have already setup asterisk and have a configuration just need to be sure

...Drop-down menu from which to choose the customer - Drop-down menu in which to indicate the name of the practice performed - Time taken - Small description (field with not required asterisk) - Self-compilation of the data in which it was made 7) Drop-down menu from which to choose the Cost of practice ADMIN 1) CRUD company 2) CRUD user 3) Dashboard with reports

Hi Arshad, we're looking for someone skilled in FreeSwitch or Asterisk to build a ringless voicemail system. We operate [login to view URL] which is a popular ringless voicemail platform in the US. However, we use other provider's APIs to do the delivery. This isn't cost effective for us. We need to operate our own delivery infrastructure. Is this something

We are looking for a technical person who has good telecom experience and can help us developing ringless voicemail broadcasting system. There are very few companies are doing this and we have no idea how do they do that. Here is detail about companies that provide ringless VM : [login to view URL] [login to view URL] Http://[login to view URL] You can read about it and tell me if you can do it....

...OVERVIEW I have to shift my existing asterisk vicidial server files ( my dial plans, modules , database ,IVR , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to a better hardware /server with Xeon processor 3.2 GHz.& 16GB RAM NEW VICIDIAL server details & its configuration ● Centos7, Asterisk, FreePBX & VTiger is already ...

This current project shall be awarded to individual who can install IP based PBX alongwith open source billing module ([login to view URL]). The IP PbX shall be asterisk based on linux platform. VPS for installation will be provided. The payment terms include 50% payout at the time of 50% work completion and rest at

After login, receive message "cannot connect to asterisk"

Dear all, I am seeking someone who can write a script that will install LAMP, asterisk, a2billing 2.1.1, and FreePBX automatically on CENTOS 7 by firing one or 2 command line(s).

Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

I have requirement of asterisk PBX developer

...(Cacti, mrtg, Nagios, PRTG, OpenNMS) b. Server Management Tools (Hyperic HQ, Zabbix) c. Backup Management Tools (FreeNAS) d. Inventory/Asset Management Tools ([login to view URL] with GLPI) e. Visitor Management Tools ([login to view URL]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools

Hi -I want to install freepbx and configure termination gateways on my server using vpn. I already installed asterisk and a2Billing -configure a connection with clients

...MT7628N CPU Cores: 1 CPU MHz: 580 Flash MB: 8 RAM MB: 64 server send call to router >>router send to voip gateway we install our program on router and our router OS is openwrt like server to router resisted and router to Voip gateway all connection By SIP asterisk server to creytiv/re router then creytiv/re to VOIP gateway some one already

Help programming sip endpoints, applications voice & sms in [login to view URL] I need to connect my [login to view URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [login to view URL] Prior experience

im looking for asterisk pjsip developer for support our system

In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

We have 2 platforms: 1. Freepbx 2.11 and Asterisk 11 2. FreeBpx 14 and Asterisk 16 CallBack makes callbacks simple. When your agent receives a callback request, they will see all of the caller’s information in the screen. They can click one button to accept the callback request and [login to view URL] will automatically connect them to the caller. This process

hi have a ip500 with ip office ver 10. with analog phone line and zip trunking were install some time back 40 constant calls only can see 24 of them 10 1616i phones with bm32 module so I need some body with Experince in avaya platform also will like to connect a remote phone,voice mail

need someone to fix the extension file on the asterisk script

i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

We're looking for someone who's able to get these key features to work between the 3CX PBX and Odoo 12: [login to view URL] If incoming call hits the 3cx server we need to open the contact from odoo linked to that phone number. Like here [login to view URL]

I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

we have a java script which is connected to asterisk we need someone to help change the settings

I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

I have setup an Asterisk Cloud deployment and need some help tweaking the final steps of the setup

...generate a voucher of the customers choice, enter the customers number, then click buy. The consumer will get a SMS containing Voucher number. (SMS sent out via local Asterisk Server) Major features for the initial design include: ● A minimal UI than allowed vendors to generate a unique token for client use. (UI needs to be made modern) ● App will

I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc

I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

...and connects both parties 2) Ability to add an audio file message (in wav or mp3) to the placed call, greeting both parties upon pick up 3) To store voice messages on my server (in wav or Mp3), email notification when message is received and ability to ftp download message Some of the existing systems: [login to view URL] , [login to view URL]

We are looking for a company who has 24/7 server administration. We have between 4-6 servers running at the same time, and we are looking for a new company, who can be in charge of everything concerning the servers. It is important that there is 24/7 support, and that there is system admins attached, if it is not only system admins. If every case goes

I would like create agent survey on Issabel/Elastix. Need modification in UI as well like adding menuitem under call center reports section to show Agent survey results page within elastix/issabel UI.

we have 2 extension files we want someone to copy the dial out routine from one to the other

I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

after cloud server down we have: [2018-Dec-04 10:22:27] [CRITICAL] (admin/[login to view URL]) - Connection attmempt to AMI failed when we try: [root@pbx]# fwconsole restart Asterisk not currently running Running FreePBX shutdown... UCP Node Server is not running Running FreePBX startup... Starting Asterisk... [--------------->------------] 13

...( can use odoo or erpnext or any other erp system) system can be integrate with doormakaba devices, for door access cards xpossible devices ( for wifi access) and PBX ( panasonic pbx ) we are looking for odoo or any other erp system which ALREADY have these options. please don't bid if don't know the working of these devices. i am not looking for

...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

We are looking for Java developer who can develop CTI & IM desktop software for our asterisk based PBX. for instant messaging we are currently using Openfire server but other XMPP servers are also welcome. We already have design and initial code to start with. Details of project will be shared on request.

...notification on my TV when incoming call, i want to see the caller id on my TV) 4 - [login to view URL] (i already have media plex server at home, i want to manage it from my dashboard and control player and choose from library and stream it on one of my chromecast) 5 - [login to view URL]

...merged both at the moment but needs the integration for call flow like: [login to view URL] 1. session linking in gui. 2. having use proper pbx in gui 3. call not approaching with proper pbx handling. 4. having issue in core integration gui. like provider, active calls, gateways etc...

I need code line examined, and a custom report built in Java and possibly PHP. This custom report needs to be built around an existing call...built in Java and possibly PHP. This custom report needs to be built around an existing call center platform and the report will need to seek and parse data from an existing PBX/Call Center Platform/FreePBX.

Hi Prostitov, I need help with our PBX system. Will you be around in about 5 hours from now?

now i have one like when a calls come to server [login to view URL] [macro-answer] exten => s,1,AGI([login to view URL],${ARG1}) and php agi write all incoming calls to database and web php read from DB but i dont know where is prob i saw some people they read active call every second same like console CLI real-time i need something realtime like it have

PLEASE DO NOT OFFER TO MY PROJECT BEFORE READING. IF YOU READ THE PROJECT PLEASE TYPE "YES I READ YOUR PROJECT" ON YOUR OFFER, AND D...will not ask Twilio for the same number if it is already asked before and already the database. P.S.; definitely we do not want to use the asterisk. We would like to run it on VoipSwitch. please do not offer asterisk.