We are a landscaping business and need 2 different MP3's recorded for our voicemail greetings. We have the wording prepared, they just need to be recorded. 1 greeting is 52 words, the other one is 75 words.
We need installation of newfies dialer, to get 2000+ concurrent calls from each box, it should be build on VM enviorment. Also asterisk amd module and beep detection modules need to be install.
Hi Everyone. I need your help on the following : #1. Configure SIP trunk #2 Call forward to mobile phone numbers with a strategy in Freepbx or 3cx, Asterisk or other VoIP services . A strategy is including Linear, Fewest Calls strategies. The calls using the SIP trunk.
...and it includes beautiful animations. UI should be based on Bootstrap - Fully dynamic content that can be controlled through admin control pages (CMS). - Adding/deleting/modifying properties and their categories and features in an easy and flexible way through the CMS. - Featured property attribute, and a section to list all featured properties. -
Need a very experienced asterisk developer for robot calls. Existing system is preferred. Using my individual asterisk server, sip accounts, multiple channels - has to be integrated to basic crm system. No twilio, no nexmo, no any 3rd party app. Any general bid ("I'm good in wordpress and shopify etc." OR " I can adjust twillio, nhx or similar") will
Hello, we need asterisk AMI script (syntax) for yeastar PBX we need functions must work via triggering AMI commands described and tested : 1. hangup 2. mute 3. attended transfer 4. hold please only serious freelancers with experience. Please be aware that Yeastar PBX has limited manager commands
Design Company products flyer, 1. Total pages is about 40pages 2. It should be in English, and English text is ready but need some modifying. 3. It should be in European and American styles .
We need an interface software for a FreePBX phone system and a hotel PMS system like Opera, Fidelio, and other hotel PMS systems. The key features we are looking ...if this is something that can be done. Features needed in this software: Class-of-service setup Extension update with guest name Room status clean/dirty Call accounting Voicemail cleanup
I have an Asterisk-FreePBX System with multiple disks that needs some fixes. If You are a specialist in this field, lets talk. --- This is not for people who think all the answers are on the Internet ! This is for experienced specialists. Requirements: Asterisk, FreePBX, SSL Certificates (Letsencrypte), Apache, multiple disks on system, web dev, PHP
...Design: Stock imagery is OK (acquired under your existing license). - Typeface: We like a simple easy to read font like Calibri or something similar. - Colors: We are open to modifying the colors of our logo. Although there are only 3 primary colors, we like the swirl with 4 (or perhaps 2 colors). Once we select final colors, we’d like them to match our
I need a logo for my photography website at [login to view URL] I a...want a logo that captures my brand as a trendy modern photographer. I have an existing color palette (#F69679, #FCDAA9, #F9CBD7, #9D5B6C, #F46E49, #292929), but I'm open to modifying it. I have an initial idea (a monogram of my initials), but I am open to alternative suggestions.
...programmer who is familiar with Voip Systems and with the following : (1) SIP: (1)(1) Setting UP SIP Server. (1)(2) Operation of SIP Server. (1)(3) SIP Express Router. (1)(4) Asterisk. (1)(5) VOCAL. (2) Protocols: (2)(1) H.323. (2)(2) SIP. (2)(3) Media Gateway Control Protocols. (2)(4) Proprietary Signalling Protocols. (2)(5) Real Time Protocol & Real
Modifying software to fix errors, adapt it to new hardware, improve its performance, or upgrade interfaces. Directing system testing and validation procedures. Directing software programming and documentation development. Consulting with departments or customers on project status and proposals. Working with customers or departments on technical issues
I ...Communication Built-In (TwiIio , CaIIraiI, cIicksend) * Automated Followup Sequences * Contract Generation and Sending * Send Physical Documents * Integrate website forms. * Ringless Voicemail Drops Do not apply for the position unless you speak English at a high level. Do not apply for this position unless you have done this exact stuff before.
Modifying software to fix errors, adapt it to new hardware, improve its performance, or upgrade interfaces. Directing system testing and validation procedures. Directing software programming and documentation development. Consulting with departments or customers on project status and proposals. Working with customers or departments on technical
Hello I have Asterisk dialer and I need to set up speech to text transcription (ONLY) I use to use to use IBM watson api for this, but it has become too pricey. it is 1 Min length audio of ivr recordings each. But total millions of files. every 2-3 months 7 million 1MB, 1 Min audio files.
Hi We are looking for a freelancer experienced in Asterisk. Current developer works at another job, so you will work with me for a long term if you want. hourly rate is 25~35. 40 hours per week Thanks Anthony
...connect those calls to IVR Instruction must follow below requirements: 1. FreePBX version is 188.8.131.52. Instruction should be based on FreePBX correct version but not on Asterisk. 2. Instruction may instruct using CLI or exactly where and how to modify FreePBX files 3. Logic of FreePBX related with these 2 tasks 4. Project will be considered as completed
Need someone familiar with the chemistry and components of exterior latex paint to aid in modifying existing solutions to fit my specific needs for a custom project. Looking for someone to ultimately create a specific hydrophobic coating system. I’m looking to create a clear super hydrophobic coating system than can be applied to exterior latex paint
We have the issue in the production FreePBX 16/asterisk 13. After some uptime or always after applying changes pjsip endpoints go to unavailable state all together. The only way to resolve is to competely reboot the pbx and to open softphone once on the end-user side. The issue doesn't affect regular sip peers. However, our requirement is to use pjsip
Hello, I have asterisk - Elastix in my office and Yeaster S20 in other location connected over Sonic Wall VPN, i created SIP trunk between both and registered on both side. i'm able to make calls but one way Audio. i need troubleshooting in configurations. on my Office - Elastix 2.5 - Sonicwall TZ400 Other Location : Yeaster S20 - Sonicwall Soho
...number is active, tey must configure it>> client logins>> enter a phone number to forward their call to (when no sip client is registered)>>client enter email address to send voicemail messages to>>client set a password for their sip account, with their assigned DID number as a username>> client saves settings then they receive an email with links to the
Actualmente tenemos ya funcionando una centralita Asterisk con FreePBX y queremos implementar un sistema de encuestas telefónicas para valorar la atención telefónica que damos a nuestros clientes. El funcionamiento deseado podría ser UNA de estas 2 opciones: - Opción 1: tras hablar con uno de nuestros agentes, la llamada pasa automáticamente