Digitmap freepbx polycom jobs

Based on our conversation the following configuration changes are to be made to your FreePBX system: Create extensions for softphone and Cisco SPA8000. Configure extensions on applicable devices. Create ring group for inbound calls, and reconfigure routing to first use ring group then go to existing IVR. Diagnose and remediate issues with email and configure Fax line.

I require an extremely knowledgeable Python developer with very good experience with Google API usage (TTS/STT/NLP), OpenAI API usage, FreePBX usage (ideally PJSUA/PJSUA2 to integrate the Python script with an extension on our FreePBX server). This will be a long term project with consistent compensation. Do not waste my company's time if your experience level is not superb. This is an enterprise level application and there will be requirements to sign an NDA prior to work. The selected developer will be utilized for potentially 6 months+. We will have an in depth discussion regarding the project.

Hello, i use this software from github. My settings are: Outgoing calls use extension 333 which forces calls to "opt...uploaded contact list, script start calls and connects to the IVR.) The problem is: Have current settings for outgoing calls via TRUNK (DongleX). I want to be able to call from another extension, which is instaled in freepbx and it works normally. $callFile = "Channel: Dongle/dongle1/$numbern"; $callFile .= "CallerID: $caller_idn"; $callFile .= "Context: callblastern"; $callFile .= "Extension: 333n"; Task: Set outgoing connection via another TRUNK (not Dongle). (Missing - correct outgoing "context" and setting in FreePBX) I am looking for someone with experience. freepbx 15, aste...

Job Description: Have installed the latest version of FreePBX on new hardware and Cisco SPA8000 . Requiring a talented Freelancer for this system to: Connect to Yealink T28P x 8 Connect to 2 physical PSTN phone lines 1 Fax line via Cisco SPA8000 Route calls from Cisco SPA8000 to FreePBX A separate inbound queue for each phone line Outbound line selection based on grouping Configure outbound email notification! Headset programming Set up / Configure Sangoma Softphone App Any other minor tweaks. Immediate start 5/1/2023 10:00am EST! Skills: Asterisk PBX, Linux, PHP, VoIP

Hi there, I'm looking for an experienced freelancer to help me with integrating an existing VoIP platform with an existing CRM. Voice over internet protocol(VoIP) is an ever-evolving technology that can greatly improve the way businesses communicate both externally and internally. I will be using FreePBX/Asterisk as the VoIP platform, and amoCRM/Kommo as my Customer Relationship Management (CRM) system. I need the following integration: VoIP Integration to allow real-time communication via telephone, Sales Pipeline Integration to track sales leads in my CRM,Click2Call, Incoming Call Card, Creating a Contact card, Smart Forwarding, Call Results card, Call Logging, Incoming Leads, Call List, Built-in call feature (WebRTC). All of which has examples on

Required knowledge of Linux (CentOS), Asterisk & FreePBX. -I need a command to delete files older than 32 days in the Asterisk recording folder and voice mails. /var/spool/asterisk/voicemail/default/ => file pattern msg????.???, recursive /var/spool/asterisk/monitor =>all files, recursive -I need to create a cron job to execute the deletion weekly. The user (meladmin) I use to manage my server is in the following groups: meladmin, wheel and asterisk. But I cannot modify or delete files created by Asterisk. -I need to be able to modify and delete asterisk created files via WinSCP. How do I modify the meladmin user? -I’m trying to use the Voicemail blast. I can call it from an IVR, but the extension cannot access it. How to make it available from the extensions? ...

...data network; Expert level experience with IP administration and management – also have awareness of the latest technology trends in Networks (Video, Data Circuits) Expert level experience with Cisco Switches and Routers (experience with Juniper platform acceptable); Cisco – CCUM (Cisco call Manager), SIP, DID Lines, Knowledge on clear pass server would be added advantage. Video- Knowledge on Polycom video device CCNA Voice, CCNP Enterprise (Essential), CCIE Enterprise (Would be added advantage) Expert level experience with designing with and managing protocols such as OSPF, BGP, ISIS, RIP,MPLS, LDP, Experience with Zhone, Occam (Calix), Hatteras, Adtran, and Actelis (1.) Incident Management + Change Management (Planning & Execution) On - call Escalation Should ha...

I am searching for an experienced Freelancer to investigate why my web interface is not working for FreePBX and A2billing. My server is running on RHEL/CentOS, and I am not sure if the web interfaces are producing error messages. Interestingly, I am finding that other parts of the system are still functioning properly, so I need someone to debug why the two programs are not working. If this sounds like a job you're interested in, I would love to hear from you!

I'm looking for someone with experience in configuring a GOIP GSM gateway to work with FREEPBX. The role needs to be completed quickly and efficiently, and the results must reach the highest standard. If you feel you have the skills and expertise to complete this project to a high level then please send me your application so we can discuss further.

We had 2 FreePBX systems got hacked over the weekend. We have to find how they managed to get in and fix the issue.

I have a FreePBX instance and an Avaya IP Office system and am looking over time to begin moving my voice service to the FreePBX but want to do it over time as time permits. I'd like the FreePBX to act as the Primary PBX trunking calls to Avaya and being able to pass calls between systems.

...your previous CRM projects and modifying it to suit my needs. Of course, there are many open source CRM solutions available but I really want to build my own. Many do not have phone integration which is a must as it's the most difficult aspect of managing communications within a medical office. I have a number of suggested solutions that I have yet to fully explore (Zoho, Salesmate, CallRail, FreePBX, etc). I'm told that Ring Central has a strong history in the medical field and that they have the best API on the market (per an independent person who has worked with several). Furthermore, they apparently have an almost "ouf of the box" solution though some additional development may be required. But they have fees and perhaps others would be better. So...

Repair , delete zulu and many other modules

Need to add Staff Freepbx SIP Extension in Laravel Website Staff Login to dial customers after the login web browser

We have FreePBX running on AWS and want to move to a fresh new install of FusionPBX on our other AWS account and need an expert to lead/ guide us, specifically: 1. Stand up the new instance of FusionPBX (current version) - with support of our sysadmins and create one tenant with our sysadmins to learn/test with 2. Document the procedure to migrate (noting configurations/SIP changes etc.) (we will configure the extensions - you will sanity check), cutover, and validation-test. 3. Support our sysadmin to perform the migration & cutover, and if needed the fallback The FreePBX configuration (for scope reference only) is attached.

We provide cloud-based VoIP services for a small hotel in the US using AWS as our cloud service provider. We have two AWS accounts, one named “Legacy_AWS”, the other “New_AWS” Legacy_AWS has a server, named Hotel_FreePBX_13, running FreePBX, configured, and providing service to the hotel. We need to migrate that server from “Legacy_AWS”, the other “New_AWS” and would like an expert to lead the process and be assisted by our two sysadmins. Overall we will make preparations for, and then: 1. Execute a pre-migration performance check on the Lakeside_Inn_FreePBX_13 running on Legacy_AWS 2. Cutover 3. Execute a post-migration performance check on the Lakeside_Inn_FreePBX_13 running on New_AWS Then if needed: 4. Fallback ...

we are looking for voip expert to setup a system that will handle call termination must work with anydesk

Setup FreePBX on Hetzner with instructions and make two phones talk to each other.

Need the following 4 customizations 1. After queue call completion it will dial another extension for survey 2. After call hangup agent will be paused automatically 3. Get Dial and Hangup event from agent outbound calls via call to queue 5. Soundboard , so that agents can play pre recorded files at any time of the conversation

I have 2 asterisk servers A and C A is running on Centos 7 and has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current configuration is A -> B > C -> outside Trunk We need the following to be done 1. A SIP remote extension to log in to A 2. A passes the registeration via B (OpenVPN) and register as an IAX2 extension in C 3. SIP remote extension made the call, A will translate the call from SIP to IAX2 Another thing, you would need to access using Anydesk via my PC to access both A and C due to VPN. Thanks

I need someone who experience in Linux and script development knowledge for Asterisk /freepbx patching. I will integrate existing asterisk PBX to Kommo CRM. The Asterisk/freepbx V18 PBX and kommo are ready, but don't know how to run the script development as below link:

I need an expert in installing FreePBX system on our server. I have specific requirements and I want these requirements to be fulfilled before setup. If you think you are the right fit, please let me know – I am looking for someone to start this project as soon as possible. Thank you for your time!

We are looking for someone who can integrate Asterisk in our Laravel system. For each incoming call, a pop up should open with the details matching in our existing database. For outgoing calls we should click the phone icon and call should be redirect to the customer number. You should have prior experience in Asterisk /FreePBX integration with Zadarma or Plivo.

Hello, We have server on which is installed FreePBX. We want to buy Yeastar TG100 GSM and Cisco SPA502G. We need to configure this 2 devices so when call is made to GSM SIM cart that is in Yeastar the call to be trough VOIP to Cisco

I'm looking for someone experienced in setting up a SIP trunking configuration with the Freepbx 16 platform and Twilio. Specifically, I'm seeking assistance setting up an IP-based Twilio SIP trunk configuration. Knowledgeable in settings and settings optimization is a must. Additionally, I'm not looking to setup Voicemail at this time so Voicemail setup experience is not required.

Looking for someone who can configure auto dialer for freepbx We already have it working but now it dials the customer and when that is picked up, it dials the agents. It should start dialing the agent the moment it starts ringing on the other end

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. this module had to be compatible with latest freepbx 16 version. - Freepbx Asterisk distribution : - superfecta module to work with : fixed price : 100 $ - ASAP within 1 day.

The project is a design and implementation of a doorbell using raspberry pi4 with python using asterisk (or another open source), zoneMinder (using ip camera), with 2 android mobile phone, at deferent flour. I need only the CODE of the asterisk (or another open source like: freepbx or fusionpbx), in orderto call and listen rasberry pi4 and android mobile phone, (flour number 1 or flour number 2). I also need a intercome between the 2 flour from each mobile android phone, at deferend flour.

Hi, Freepbx server up & running fine on AWS 13.51.***.*** Skyline GSM VOIP gateway 192.168.***.** Configure require to communicate FeePBX with Skyline by using anydesk & Add Sip Add Trunk Add Inbound- info available in excel Add Outbond- info available in excel Quick completion require

Hi, Freepbx server up & running fine Skyline gsm voip configuration already working on cloudfonica but i need to use now freepbx Need to configure skyline with freepbx by using anydesk Sip Trunk Inbound info available in excel Outbond info available in excel Quick completion require

Caanot update fwconsole ma update userman An exception occurred while executing 'ALTER TABLE userman_password_reminde r ADD PRIMARY KEY (id)': SQLSTATE[42S02]: Base table or view not found: 1146 Table ' _password_reminder' doesn't exist

Hi, Freepbx server up & running fine Skyline gsm voip configuration already working on cloudfonica but i need to use now freepbx Need to configure skyline with freepbx by using anydesk Sip Trunk Inbound info available Outbond info available Quick completion require

We use Google workspace at work, so all' our contacts are stored on Google contacts. We use freepbx as our pbx and I would like to be able ti have both connected to see the name of Who Is calling via freepbx querying g contacts.

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. - Freepbx Asterisk distribution : - superfecta module to work with : price : 50€ fixed - ASAP within 1 day.

seems freepbx not connect to AMI but seems config file is ok, something wrong with this server, and need fix start by log. some days ago server work perfectly, need help with this.

On our IncrediblePBX vps : add contacts db source and make connection with growl to get customised notification based on SIP line, and caller details if found in db. price : 50€ fixed

Look forward freepbx expert to verify existing installation, and correct if any problems. also guide basic settings. Thank you

1/ install/setup Freepbx on AWS EC2 instance 2/ config Twilio SipTrunk 3/ Config 1 Extension

Client requested to install FreePBX on Raspberry PI, installation was completed through Anydesk in 2 hours, video of the installation has been recorded.

i need some python scripts for my freepbx asterisk voip system

Well established online "advice site"- needs to complete our Asterisk PBX system build: Minimum of 3 years previous experience preferred. Proven Experience with Asterisk 13 or later, Apache, Linux, PHP, (Laravel) and MySQL required. Knowledge in TCP, SIP, RTP, UDP protocols. Experience and advance usage of FreePBX or Elastix. - Dial-plan programming in AGI, AMI, REST / ARI ( in PHP ) - IVR programming in AGI, AMI, REST / ARI ( in PHP ) - SIP trunk & Gateway integration with third party VoIP providers Experience in REST API development and websockets. Knowledge in & paypal payment gateway Knowledge in Git. Strong troubleshooting skills a must

Recently I formatted hard disk on server running VMWare ESXi I recently formatted esxi and now have two vm. Need to configure as having network issue. I did ...onlink command, if-uo etho in one of vm, but still destination host unreachable. also do you have experience with bgp quagga, debian, centos, ispconfig, freepbx? I also have backup of VM in tar file. I have created two VMs running Debian, but now need to restore the tar files to those VMs. I went till cd /vmfs/volume/Datastore1, but not sure which command to run to get to the actual VM command line or restore the tar file to VM. Please note no credentials are shared and access provided over anydesk or teamviewer

Hi; I am a private IT enthusiast, who is interested in learning more about IT. I have recently come aware about Asterisk, Kamailio and Opensips. I would like to build a lab using virtual machines to see, how all of this works. What I am looking for is a k...this works. What I am looking for is a kind of tutor, who could guide me through the steps to build this lab and get it working, while explaining to me tha essential things that I need to know. I am not a professional, so I do not need much details, even though sometimes I would have to get some. If anyone is interested in helping me building this lab once with Kamailio/Issabel and Kamailio/FreePBX and also using Opensips - connect it to Jitsi or MS Teams, do not hesitate to contact me and tell me, how much would it cost. ...

I am looking for setting up dinstar gsm gateway for incoming and outgoing calls, also enable webrtc for extensions to make and receive calls. My budget is 50$ for this ,please do not bid higher.

...Dongle Panel Function : web-base hosted VoIP to GSM gateway is accessible from anywhere on any mobile device, it helps to terminate/originate SIP calls traffic using 3G Modems/Dongles and an active USB hub connected with your Linux PC. With the features of exquisite enclosure and high performance, specially designed to maximize cost . Dongle : meaning Huawei usb modem with sim card attached to Freepbx or issable End-User • Same Panel (As Attached Pictures) • Add Call Feature via panel • For all functions I supposed to choose which dongle i need to use . Admin-User • Control All users (Add,delete,rename,change password,list of all users) • Payment method : PayPal & Stripe (2 options will be monthly or one time

I need OTP voip number that fowards the SMS message via Email. The OTP must work with Amazon Seller Central, Walmart, HomeDepot, and many other vendors. We've found some VOIP services work 90% of time but then didnt work with formatting for say, Amazon. Just need a "soft" solution to receive company OTP SMS messages. We do have Twilio and with Freepbx sofftwares. We do have a "OTP NUMBER" that would need to be transferred to the "VOIP SOLUTION" from Verizon Wireless.

What is expected • Availability outside normal business hours on demand. • Ability to create and maintain system documentation (policies, diagrams, etc.) • Strong knowledge of Windows Servers, Unix, and network/web/core subcomponents. • AWS, GCP backup, recovery and health monitoring practices. • Experience with PBX systems (FreeSwitch, FreePBX, Asterisk, etc.) What is good to have • Experience in managing Database servers (MsSQL, PostgreSQL, etc.) • Knowledge of scripting languages (PowerShell, bash, etc.) • Understanding of TLS/SSL and certificates chain use/distribution What is not required • Customer support • QA (Testing, bug tracking, etc.) • DevOps • User training Expected employment type: • Full-time (9a.m. ...

Can not send “@” in sms by Dongle Usb Example: I send: test @ I get: test ¡ What im missing in dialplan ? [textmessage] exten => 111,1,NoOp(SMS receiving dialplan invoked) exten => 111,n,NoOp(To ${MESSAGE(to)}) exten => 111,n,NoOp(From ${MESSAGE(from)}) exten => 111,n,NoOp(Body ${MESSAGE(body)}) exten => 111,n,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)}) exten => 111,n,MessageSend(${ACTUALTO},${MESSAGE(from)}) exten => 111,n,NoOp(Send status is ${MESSAGE_SEND_STATUS}) exten => 111,n,GotoIf($[“${MESSAGE_SEND_STATUS}” != “SUCCESS”]? sendfailedmsg) exten => 111,n,Hangup()

We want to outsource the pre-sales area, in the audivision systems division, with demonstrable knowledge in AMX, Creston, Control4, Polycom, Extron. The project is based on reaching an agreement for each quote made, which includes: economic proposal, design and scope. They must know how to handle Visio and Autocad.

We are cur...of AV experience 5 years managing a video network environment Experience with Polycom RMX Experience managing CMS, and TMS Must have a current or working towards Avixa CTS certification and maintained during employment. Exceptional customer service abilities. Superior oral and written communication and problem-solving/strategizing skills. Willingness to learn and adapt to new systems and procedures/stay current with AV technology. Flexibility in daily work hours to meet the conference clients' needs including weekends. Knowledge connecting to different Unified communication platforms including but not limited to Cisco Web Ex, Poly, Zoom, Blue Jeans, Skype Technologies: MS Suite, Zoom, Teams, Cisco CMS, TMS, Polycom RMX, RSS, Crestron, Biamp, other Unified Co...