Conference Bridge With Raise Hand Feature in Asterisk / FreePBX. I need for a client as follows; Number to call in, reaches announcement with instructions. Then goes to conference line where an instructor and moderator are talking. No one else can talk (everyone else muted). All the rest of the callers in the conference that are listening should have an option to press a DTMF key, that basically raises hand and that places the caller in a queue but DOES NOT pull them out of the conference ONLY when the operator in the queue is available (not on a call). Needs someone with extensive Asterisk development skills and need to have basic FreePBX skills
Conference Bridge With Raise Hand Feature in Asterisk / FreePBX. I need for a client as follows; Number to call in, reaches announcement with instructions. Then goes to conference line where an instructor and moderator are talking. No one else can talk (everyone else muted). All the rest of the callers in the conference that are listening should have an option to press a DTMF key, that basically raises hand and that places the caller in a queue but DOES NOT pull them out of the conference ONLY when the operator in the queue is available (not on a call). Needs someone with extensive Asterisk development skills and need to have basic FreePBX skills.
Necesito persona con conocimientos de centralitas asterick, montada sobre una raspberry 2, freepbx concretamente, configuración de extensiones, ivr, contestador, franjas horarias.
Hi. I have a new fresh install of freepbx 13 with asterisk 13 and cdr dont show calls. In folder monitor are the files, but dont show in CDR Report.
I need astersisk to be installed . using freepbx. and also setting up some DID numbers
The purpose of this document is to outline the needs of CQ Simple for the phone provisioning software that is needed. All the OEM phones will be from Atcom. Currently the model is the CQ Simple CQ400 (Atcom Rainbow R3). There will be other models to come so the software must have the ability to add new models as needed. Currently the Nimbus Network Platform is running an OEM of Freepbx. The underlying OS is Centos 6. This is a LAMP installation. All communication with the phone and the phone system will be using the tftp server on the Nimbus platform. At a minimum the software must provide the following: 1. Provide a means of programming all the buttons on the phone from a webpage on the platform itself. 2. We must have the ability to create / edit different “Tem...
install freepbx and asterisk 1.8 on dedicated server
As ...need is to identify “When” a live-person is on the line. We prefer to adopt/embrace existing software packages and make customization and not to develop all from scratch. Our cloud system is based on AWS and Linux environment, PHP and AGI. We prefer development in PHP but other scripting or code languages are ok. Larger packages that may be potential to start working with are Asterisk, FreePBX or FreeSwtich. Any other free software platforms are fine. It is possible to use some transcribing services as a feature to detect a live-person. Potential candidates can be CMU Sphinx, Dragon Nuance, Google Speech API, or IBM Watson Speech-to-Text. Other packages are welcome. We need to hire several contractors, and this small task can be a good tool to identify ...
I currently need integration built with Asterisk Phone systems specifically FreePBX. This integration should look up the caller ID while the screen pop displays customer info from a database accessible via API calls. Reference Guide: NOTE: (Should be built to work with other API Calls with other systems) Once a successful match has been made options will show up, New Ticket, or Existing Ticket (via lookup of matching open tickets for client). If creating a new ticket by selecting the button we will popup a screen to make notes, and start a timer to record the start time of the call all the way to hang up. If existing ticket then we just append a new time charge with notes to the ticket. Other integration's would be nice too.
SIP, Asterisk and FreePBX Administrator required for remote troubleshooting and support. We require an Asterisk admin to remotely handle a few tasks: 1. Set up Caller ID for different Inbound routes. 2. Set up new SIP extension in ring group. 3. Set up new ring group. 4. Troubleshoot Yealink T22 ringing issue.
Install a 4g lte modem huawei E8278 on freepbx. OS --> centos 6.5
We have Cisco SPA phones and need a CGI XML Application for Asterisk that will display the parked calls details per specific parking lot. We were successfully using this script until we had to create an additional parking lot. We only need the group of phones belonging to the specific parking lot to display calls being parked for it. Unfortunately we need this up and working for tomorrow morning.
I'm having some problems with my asterisk configuration, I need someone to answer some questions and fix some config files
Install a 4g lte modem huawei E8278 on freepbx. OS centos 6.5
Need someone to finish to set up my freepbx to work with Twilio. Everything is set up (trunk, inbound route, outbound route and extension) but it seem to did'nt want to connect to extension.
Existing FreePBX stop working suddenly. Need to fix asap
I need configure my PSNT Line in FreePBX (Asterisk) using a Lynksys Spa3102
I have installed asterisk.. the problem is freepbx is not compatible with php 7. to install it somehow and dont touch php itself. dont have to delete or reconfigure apache/php/nginx/varnish in any way.
Need to get my freepbx system to seamlessly integrate into a galaxy lightspeed 9.7.2 hotel pms system
sir, plz i humblyy request u, plz help me .. i am installing freepbx, but i am getting error. plz solve my i am installing freepbx in vmware and in a seprate system.. for both i am getting error .. ! can u plz fix my issue ???
I have a working local voip server and I need to move it to an online freepbx server, I need to connect the online server to the local server to make it work, more details will provide in chat, please bid only if you have experience.
I'm lookin for someone who can set up and config my FreePBX VPS to work on Plivo Trunk (Send and receive call) The FREEPBX is already installed and configured, the only thing left is to config Plivo with freepbx thank
Hi, We are running FreePBX and Asterisk server. Our customers use soft phones to connect to our PBX. We want to start charging customers. We want to provide calling service and network access service to paying customers. I read that to provide Authentication, Authorization and Accounting for network access service like Wifi, i need a radius server. I read that to provide accounting and authorization service for Asterisk, there is A2Billing. Would need your advice on which software to use if we need to provide pre-paid calling and pre-paid Wifi access service. Would also need you to install for use the appropriate server side software and provide step by step guide. We will provide you with Amazon Web Service cloud based EC2 servers. They could be running Ubutnu o...
FreePBX inbound DID setup. Hello, I have a new FreePBX setup and a DID number and I cannot get the DID number to work unless I allow any connection to come through. --If I allow any connection to come through, the call comes through. --If I disable this option, it doesn't and when I call in to test, I get the default ivr error message: "The number you have dialed is not in service". I suppose I am not routing number 5116429950 properly in Inbound routes , thereby my FreePBX is playing that default IVR. I need to correct this. My product and hosting are as follows, I dont know if this helps: DID NUMBER Web: PBX PRODUCT Web: Product: FreePBX Hosting: SETUP FOLLOWED Thank
I need a asterisk programmer with good experience in freepbx and other managing tools to develop IVR
configuer installed voip Astersik software to set up a phone system using 3 analoug lines to 6 IP phones - setup system, extension and users IVR etc
Install and configure with my requirements Asterisk/Freepbx in Italian Language on Linux Server
Major Tasks are: 1) Create, configure, install and show us how to implement the webrtc click2call button on our site. Button will connect to our Asterisk PBX. 2) Install and configure OPUS on Asterisk to accept this webrtc codec (in passthrough mode since asterisk doesn't transcode OPUS) 3) Recommend, install, configure and test a windows softphone with Opus for our call center so that we can verify that the entire solution is working. System needs to work on all windows desktop browsers (except safari but including IE) and chrome on mac OS X A document with instructions basic on how to install and configure each item used should be delivered. This is importan if we need to re-install to new server or add new click2call buttons. Preexisting open source software should be used ...
I need you to develop some software for me. I would like this software to be developed for and freepbx asterisk. Need a way to call from vtiger crm through pbx module but without the user seeing the number on their crm screen or sip phone
need a hotel billing software for freepbx
I need an aws account setup with two instances. One freepbx 12 and clearos 7 Then I need to have a script which will start both, update couple directories and bring them down again.
Possuo uma central PABX instalada com asterisk e gostaria de fazer funcionar as ferramentas de queues (filas de atendimento e agentes), chamadas preditivas (propagandas automáticas) e URA.
What I need is something simular to this: http://wiki.freepbx.org/display/FPG/Queue+Callback+Module after a customer have called into a freepbx-queue this callback feature frees a caller's time by letting him or her "press 1" to exit the call queue and receive an automated callback when there is a free agent. That module is a part of VQ Plus Commercial Module and contains a lot more stuff then I need, and I am afraid it will mess with my dialplan and custom modules. This is hosted on a dedicated centos server. It is asterisk 13 with freepbx 12 on top. this must be done without breaking my dialplan or custom freepbx modules.
I need someone to help me configure a Cisco 2851 and FreePBX to handle inbound trunks, do IVR to cisco SIP phones, be able to route calls from reception to a GSM gateway and a searchable directory. I also need to keep the Voice and Data vlans separate but sharing physical network so that desktop PCs connect through Cisco phone with mini trunking. I want this to be done along with me so that I can learn how it works.
Hi, I am looking for a freelancer who has knowledge of configuring FreePBX I already installed it but need some help in configuration and for future will have some questions about it.
Hi ivan381eu, I noticed your profile and would like to offer you my project. As we talked over the chat .
1. system shell have option to dial by name to lookup an employee's esxtesion. 2. System shell have the capability to provide nested options and play different types of announcements. 3. Ability to change recordings based on time of day. 4. system shall have the ability to have an overriding message in the event of an emergency or disaster to announce closures or others critical information. 5. Allows customers to retrieve the information they required through commands without ever speaking withan agent, or to quickly navigation to the correct department or agent that can help them 6. Provides multilingual support for IP IVR server prompts, for automated speech recognition (ASR) and text to speech (TTS) capabilities
Hi ivan381eu, I noticed your profile and would like to offer you my project. I need a menu in FreePBX ( Various Versions ) Outgoing Route that i can upload List of the Numbers , and it will call random , I have such in the manual script configuration and want to make it more editable . We can discuss any details over chat.
Hi guys! I need to develop some kind of a system integration of asterisk pbx with our web-application. From hired person we will need exactly working system on rpcxml techs which will be connected directly to our server station. Right now we've got 2 server stations - 1st one is using for web-application, the second one we need to setup asterisk pbx + some kind of web-gui (it can be freepbx, freeswitch or anything, doesn't matter). Developing stuff is must be kind of a json functions library of asterisk pbx instructions, which will give us full remote control of our telephony. More information will be given to hired person.
I have vTiger CRM on the cloud and my Asterisk Server in my local area network, i need to use the vtiger connector to get the reports of the CDR and make call on the CRM.
We wish to have some custom work done based around call forwards on a FreePBX server. The server will be running the 'Offical FreePBX' distro - CentOS 7, Asterisk 11, FreePBX 13. However, it is running in a hosted environment and root access is not available so coding must be done using the 'Config File Editor' built in to FreePBX 13. The requirement is the "user" needs to be able to call an IVR (with PIN protection), select which department to set the call forward for, and then enter a 3 digit number which will relate to a ring group that has the external number in it. The reason for using ring groups is we want to make it easy for the user's to log in and administer the ring groups themselves. As for overall call flow, ...
Hey, I am looking for someone to assists me with the SIP configuration. should take up to 10 minutes. Thanks.
Project Description: Asterisk 11 and 13 CentOS 7x MySQL 5.5 FreePBX 13 The feature should work as shown below with documentation. Can be AGI or Dial Plan () Record all calls between agents and customers but only start recording when customer is bridged (talking) to agent. We also run “Queue Callbacks” so Exten and CID might be mixed. MixMonitor(/var/spool/asterisk/monitor/${YEAR}/${MONTH}/${DAY}/Outbound-Agent-${AGENT}-${CALLERID(num)}-${UNIQUEID}.WAV) Only record calls that originate from a Queue. To include on string Exten(Agent) Queue number, CID (Customer), Date, Uniqueid Write to DB exten => _XXX.,n,Set(DB(call_tag/${ EXTEN } ----MUST BE AGENT---- /var/spool/asterisk/monitor/${YEAR}/${MONTH}/${DAY}/Outbound-Agent-${EXTEN}
...call an agent will ask the customer if they are prepared to answer a short survey and the call will be transferred to the survey module. The customer will be asked questions via system recordings we will record and asked to press the appropriate number key to indicate their response. Detailed requirements: * Must be delivered as a FreePBX module compatible with FreePBX 5.211.65-21 and asterisk 11.14.2 * Must be configurable through a web page on the FreePBX web interface * Configurable features must include: number of questions, range of values for response * Results will be stored in a MySQL database with information on the call, the agent, the questions and the answers to the questions * There will be a web interface to display the result for each question acr...
Looking for someone with a ton of experiance with FreePBX on a command level not just GUI, our reporting is showing for example that we had 157 answered calls but yet for the day we have 250 recordings, I see a recording for a caller ID and there is a 10 min call but in all the reports the call shows as abandon. need to get to the bottom of this and make our reports accurate. Looking to do this asap
I am looking to have developed a freepbx call accounting module. It should have the following features 1. Setup rates for trunks in FreePBX. This should allow for rating numbers prefixes. The system should match the closest prefix. It should rate all types of FreePBX trunks iax, sip, dahdi and others. It should also rate based on time of day and specific increments. 2. Importing and Exporting of Rates - Rates should be able to be exported to csv or imported from CSV. 3. Extensions in freePBX should be able to be enabled and disabled whether they will show in the call accounting reports. 4. The system should be able to push out 2 reports. A summary report of all extensions enabled for call accounting and the total cost of the period selected. A detai...
We are a US based tele-health company providing educational doctor-to-doctor video consultations. We started providing this service also in China (US-Doctor to Chinese Doctor video consultation). Currently we are trying to connect to the tele-medicine department of a large hospital in China that is equipped with Polycom. In addition they have access to Vidyo. Put we are having problems to connect to their system from the US. We need immediate help to resolve this issue. This is not a programming job but rather a consulting job where you would join the conversation as one of our consultants. Thank you, B
