Asterisk ivr projectJobs
Vb6 ivr asterisk dialogic programme
Sonetel & Asterisk Configuration
a dialer for my asterisk box kasdjlasjdlasjdlasjdlasjdalsdjalsdjalskjdalskjdalskjdalksjdalskjdalskjdlksajdlasjdalsjdalsjdalsjdalsjalskjd 100chars
a dialer for my asterisk box kasdjlasjdlasjdlasjdlasjdalsdjalsdjalskjdalskjdalskjdalksjdalskjdalskjdlksajdlasjdalsjdalsjdalsjdalsjalskjd 100chars
Добрий день, Мені потрібно виконати інтеграцію Asterisk+Zendesk - бажано, щоб при надходженні дзвінку створювався тікет, ідентифікувався дозвонювач, і т.д. Чи є в Вас такий досвід роботи? Ми знаходимося у Києві. Бюджет та подробиці оговоримо.
...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
I need to include a button to call directly to my freepbx server, to sales option (ivr) customer enters number and then calling is established
Hi Arshad, we're looking for someone skilled in FreeSwitch or Asterisk to build a ringless voicemail system. We operate [log ind for at se 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
Hi We are service provider to different customer needs. One of my client needs inbound and outbound call center set up along with IVR, call recording and monitiring functionality(customer feedback system).
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 : [log ind for at se URL] [log ind for at se URL] Http://[log ind for at se URL] You can read about it and tell me if yo...
...Endpoint License (E-Delivery Smart) 9 24 CON-ECMU-SRSTGTEP SWSS UPGRADES Cisco SRST-1 SRST Endpoint License 9 25 CUE-IVR Cisco Unity Express v9.x- IVR Session (E-Delivery Smart) 2 26 CON-ECMU-CUEIVRTB SWSS UPGRADES Cisco Unity Express (CUE) IVR Licens 2 27 SISR4300UK9-316S Cisco ISR 4300 Series IOS XE Universal 1 28 NIM-4FXO 4-port Network Interface Module
...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 Installed
This current project shall be awarded to individual who can install IP based PBX alongwith open source billing module ([log ind for at se 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
...Inventory/Asset Management Tools ([log ind for at se URL] with GLPI) e. Visitor Management Tools ([log ind for at se URL]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools (Jabber) Basically the idea is to help new IT Managers and IT administrators to manage IT infrastructure using opensource tools to
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
...>>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 made it they can run more then 64calls but if you can run 32calls i am Happy need a easy way
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 [log ind for at se URL] will automatically connect them to the caller. This process
need someone to fix the extension file on the asterisk script
...Inbound, Outbound and IPPABX + INBOUND Hosted IVR ACD Skill-based routing skills Priority routing Time-based routing Queue monitoring Barge-in & call monitoring Real-time dashboard & analytics Call recording Multi-channel communication CRM + OUTBOUND Preview dialer Predictive dialer Progressive dialer IVR blast – SMS & Voice Call barge-in & monitoring
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 have installed OpenAnswer, FreePBX, and Asterisk on a AWS. We are looking for someone familiar with the software to finish fully setting it up for us. This includes changing all the default passwords. After setting it up we want to upgrade to PHP 7.2, and NodeJS 8. We are also looking to remove all test files, and default configuration files that
Hello, I am seeking a female voice over artist that can provide high quality voice prompts for a phone IVR system. The tone should be friendly and professional. 439 words total. I have provided the entire script. Each line in the attached file needs to be a separate MP3 file. Thank you
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 call out page already , need someone to change the extensions to an ivr based answer
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
...portal, 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 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
Require someone who can develop a chatbot for IVR/Call for Indian company and users. Not looking for Google Home and Alexa bots.
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
Customer should Signup with Hosted services and chose from the ready made IVR Applications or get one developed at effective cost Click to call features: 1) When number clicked on webpage, viewer types in his/her phone number and submits to call other party, Click to call calls and connects both parties 2) Ability to add an audio file message (in
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 through level 1 or level 2 support first, the...
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.
