...STH lisanslı bir telekom firmasında teknik departman müdürüyüm , şöyle bir sorunumuz var daha önceden çalışmakta olduğumuz aculab prosody s üzerine kurulu voip sisteminden asterisk platformuna taşınmaya çalışıyoruz test amaçlı kurduğumuz server üzerinde denemeler yapıyoruz, şu anda çağrılarımızda bir sorun yok istediğimi...
I need inventory software with excel import and export utility also bin configuration should be there once we upload stock and when we upload a file to check it should show in stock or out of stock
Hello all ... i have a website which works already with no issue , its hosted through VPS Linux , i need someone who is trusted & expert with high reputation reviews to configure this hosting and secure the emails with SSL certificate for both pop & smtp protocols . additionally , i need to prevent any access to my server through linux such as sudo access , ssh , root , etc .. beside t...
...initiate a transfer via DTMF tones. The current PBX (main PBX) doesn't support tranfser via DTMF. This is the tricky part: I *think* we need to put asterisk in between the main PBX and the phone. I *think* Asterisk should be able to listen for DTMF tones and initiate a transfer to the main PBX if the correct DTMF sequence is entered on the phone. I need
I am looking for an expert programmer in asterisk that can set all call rejection codes to 503 and Instantly reject all busy circuit calls without any delay by removing the played busy circuit massage, these mods needs to be done on 2 Freepbx Server running Asterisk 220.127.116.11
i got this error in asterisk and freepbx under centos "The number you dialed is not in service, please check your number and try again" im using twilio sip trunk, please i need soeone who fixed, everything i twilio is already done
Need to install metric server on existing kubernetes cluster created using kubeadm and need to configure hpa
I need to correct an existing signup form that I have been using. Right now it doesn't matter what people select, they are only able to choose one of the three options even through they should be able to check all that apply. It used to work but doesn't now. I need to fix this as soon as possible. Second issue is that we use mailchimp to do several things. 1. tell us what kind of ema...
Zoho's phonebridge plugin supports up to Asterisk 1.4 [login to view URL] With modern asterisk versions (11 and up) it's not working at all do to (apparently) java problem. And the property in the phonebridge adapter seems like it is not being set properly. With Astersik 1.8 it was working partially
OS: CENTOS minimal Topic: Zimbra Pre-Conditions: - apache web server is already installed - zimbra is installed and configured properly Requirements: configure zimbra proxy so that apache web server can communicate with that. the zimbra admin and webmail should be available via subdomain.
...on will be defined by BB1, BB2, and BB3 - We have 3 cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud
...page is to be seen or the program is to terminate. In addition, parts that have quantity on hand values that are equal to or below the reorder point should be flagged with an asterisk. Write this program as a C++ program using structures that have bound methods, functions. Write a structure, **struct card**, that will represent a card in a standard deck
Need to configure OpenVPN on Windows server. Clients need to - Access internet throught vpn - Acces files and directories like local network - Need to ping the server - Access Server through name server (DNS) and IP. Support to configure clients and confirm the succefull connection and all this itens above.
Hello, Please read carrefully this project description and apply only if you can do this job with the allowed budget I will mentioned in the end of the description. I have the IOS and Android web view uncompilled packages, and i need to configure them with my website link and make some adjustments. Are necessary the following operations: - for website integration need just to place the website l...
A company is running Asterisk with an onboarding system. We are looking for an engineer who can support Asterisk and Kamalieo system in the cloud. Engineer is required to set up an onboarding system for clients and support them on daily basis. Here is the following job scope. 1) To set up and maintain an onboarding system that allows a user to sign
hello, we recently had some work done by a freelancer on our bespoke asterisk voicemail application. however, customers are reporting various problems: connectivity, hang-ups, can't make changes, etc, etc.. : something is definitely wrong with what was done. we need an experienced troubleshooter to take a look and make changes to a live, working system
hello i want to make a dynamic filter with asterisk follow this features - the calls come from customer pass through asterisk filter to check if it's an spam call before to send to gateway gsm if the call is a spam, then the call no go to the gateway but if the call is a real, then the call pass normaly
Hi I need someone who is diligent with Voip Asterisk. We have a Voip freepbx setup on 1.8 For some reason Iptables is blocking our Phone lines. I disable iptables service. All is working 30 min later it automaticaly reenabled iptables no more Phone. So as the freepbx is old. I want to reinstall a new server. We have 2 tenants 5 trunks and 11 phones
...person to do some kamailio development for us. We would like to have kamailio look up the registrar domain and forward all registrations and invites to and from multiple asterisk servers. I think this can be done with Domain and Dispatcher module. I have written some of the config to handle registration, but when a call(INVITE) comes in it is not
I offer VOIP telephony services and development of telecommunication solutions, I need a logo and a brochure for my site currently Im using logo which is n...logo and a brochure for my site currently Im using logo which is not mine. Send sample on your propoersal based on the description of my needs This is my site [login to view URL]
We'd like to create an asterisk-based calling system for internal communications, including multicast messages and intercom calls. Incorporating a basic effective web interface, similar in effect to solutions such as Freepbx, with some custom functions and inputs.
...of our workflow (which requires internal file references) and may need to use Egnyte or local backup on a limited basis. Looking for a freelancer that can help us with configuration and act as an advisor on an ongoing basis for all of our cloud and fileserver requirements. Please answer the following questions in your response: 1. What is your experience
Need to configure 10 internal accounts on IP phones, connect 3 external telephone lines. Set the voice greeting and the voice menu according to the technical task. Implement a CDR report and record conversations.
I have 2 asterisk (FreePbx) servers up and running in 2 different locations connected with an iax2 trunk. I need help with the following: 1. Install and configure chan dongle for use with Huawei K3520 in one of the locations. 2. Dial plan between the 2 asterisk servers, depending on dialed number the call should be sent to correct dongle. I have a
my project to connect Atmel XMEGA A1U Micro-controller through SPI to two RF micro-controllers AX8052F100 that are connected to two transceivers, one acts as transmitter and the other acts as a receiver. I need someone who can write a code on AX8052F100 to do the following : the task is to write two code in code-block to access and configure the SPI, one time to be master for the first AX8052F1...
I have a thin and I want someone to help me thin Client Configuration
...from SIP to GSM - Must run on background - Must be very lightweight to run on small memory devices - Configure SIP Accounts. 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 RFC2833 or inband - Use codec
Fix an existing Asterisk PBX system with WebRTC. Right now calls don't come in because FreePBX/Asterisk doesn't register with DID supplier. To make it register, some changes should be made. The system consists of 3 servers, Apache, Asterisk and MySQL. You must be able to work with Teamviewer !
We need experienced users who have already done similar projects We need integration of embedded SIP cilent woth WEBRTC to work with FREEPBX with all functions supported : hold transfer - attended , unatennded dial etc .. link : [login to view URL] regards