Freeswitch dialplan Jobs

Our current situation We are a call centre and have been on the market for many years. We operate a classic telephone service / telephone secretariat. We are currently work...whole thing. The briefing also includes ideas on how to implement the whole thing or which existing functions could possibly be used. The operator panel should be available as a stand-alone web application and work together with the telephone system / FusionPBX / Freeswitch. The WebApp will then certainly be able to be connected to the telephone system somehow in a config file. That would at least be good. The script language on which the WebApp / operator panel is based or developed is up to you. We can install Freeswitch and FusionPBX on a Server and set up a SIP Trunk. We will give you ...

Hello We have the latest FreePBX server set with asterisk in it and Node Red to control the asterisk dial plan. Requirement is when we get an incoming call through a channel then Node Red identifies and takes over the call control. While Call forward and pre-recorded messages works fine however we are not able to get TTS based MP3 audio playback done via node red in asterisk. As per asterisk MPG123 will work however as using by nodered thus the file as to be injected via a function instead of node. Looking forward for solution to make following happen 1. Play mp3 file in asterisk(freePBX) via node red over an extension in incoming call. In a way node red will become dynamic IVR for bridge connections 2. Based on caller ID Node Red will HTTP request our CRM for CID name and announce the ...

I have recently taken this job off a developer that has been slow in completing the development. Theref...someone to make changes on the look and feel of the website, add DIDWW and DIDX APIs, Mobile Top UP API like TransferTo and Prepay Nation so that clients can add their user names and logins when integrated well, implement report to the existing SMS system as there is no report there. Test the system to make sure all components work well before we can release it. You must be knowledgeable in Freeswitch as the system is a combination of two popular programs merged together. We only want developers who are knowledgeable in FusionPBX and ASTBB and have worked on these systems before with results. We will use these systems as a yardstick to measuring how good the best candidat...

must be set on a raspberry pi 4 asterisk and freepbx with 4 dongle channels. To create dialplan to send automatic calls from cli

Looking for a Class5 Soft Switch based on FreeSwitch with Android and IOS APP.

Analisar pacotes RTP/SIP do freeswitch, usando script Dejavu Python

Hi, We are a startup and need to hire a FreeSWITCH / OpenSIPs telecom engineer to help us with tasks from time to time. We would like to work long term with only 1 developer / engineer, you must also know how to install and setup FreeSWITCH/OpenSIPs on AWS. We will pay by the hour, please send your resume or experience and price per hour you charge. thank you.

Hi, We are looking for a developer with FreeSWITCH experience on AWS . I need several things done. 1 - The last developer setup FreeSWITCH where the call recordings save in BOTH the EC2 & S3 Bucket, we just want to save them in the S3 so they need to stop being saved in the EC2 instance. 2 - I need some sort of monitor so when a registration fails or calls fail it will send us a notification, I see there are some FreeSWITCH modules for this that. 3 - A FreeSWITCH dashboard to show real time the current state and calls coming in/out, I also see some open-source ones online but am open to recommendations as long as there is no recurring fee's for them.

1. FreeSwitch in Windows platform/OS 2. FOIP in Windows OS FreeSwitch 3. SMS module 4. Conference module 5. Speech-to-Text module

Hi NetworkLab, you had done some work for me in the past, it was very high quality and good work :) Can you please send me a price quote in install one of the FreeSWITCH monitors so if a registration or phone number fails it will notify us, also, do you guys like any FreeSWITCH dashboards to monitor activity? If so which one and how much to also install that, thank you!

Hi I need a dial plan for an extension in fusion pbx where I will have an extension as follows. I call the did number that points to this extension - the system tells me to dial a phone number for identification (that is, what will be the callerid for the call - what identification will appear at the destination) - the system repeats the number I keyed and gives these options to confirm press 1 to press again press 2 - If you press 1 - please enter the phone number you want to call (destination). The system repeats the keystrokes and so on - if you press 1, the system dials using a trunk that you will have to set up (of my provider) - the call is automatically recorded. And I can receive the recording by email/ see and hear it in Fusion's interface. The budget for this is $80

Hi I need help with various settings on the subject of voip

Hi I need help with various settings on the subject of voip - required skills - diligence, flexibility, understanding, desire to progress and learn new things, consistency, availability

i need someone to install for me the best for 500-1000 cps , i need to put a blacklist of 1 millio number i tried with mysql it goes down any suggestion

Someone must get experience in FreePBX and Asterisk. Main task is making dialplan , and solve some small issues. if someone have rich experience in this field, it will not take 3 - 5 hours. Long Term Project. As based on this result, we can work continually.

Hello: We need a passionate Asterisk based Fusionpbx/Freeswitch etc specialist with programming skills. We have tested and used different forms of Asterisk and we want you to collaborate with us to make or customize a new GUI. So its not just a system installation but a new or modifying a GUI. PHP is a good program as many GUI use it including Isabelle but we are open to Python, PERL, or any other language. It can be a long term project but lets start it small :)

Hello: We need a passionate Asterisk based Fusionpbx/Freeswitch etc specialist with programming skills. We have tested and used different forms of Asterisk and we want you to collaborate with us to make or customize a new GUI. So its not just a system installation but a new or modifying a GUI. PHP is a good program as many GUI use it including Isabelle but we are open to Python, PERL, or any other language. It can be a long term project but lets start it small :)

Need Freeswitch TLS setup VoIP SIP

We already have a Voipswitch developed on Astericks and also FreeSwitch and we also have IOS and Android apps that delivers Voip calling from any country. We experience Voip blockage in some countries eg UAE and Brazil etc. We need you to be able to use Kamailio to build or create a tunnel or add mediator server that will act like a proxy that will bypass the voip blockages put by those countries . Thereby allowing voip calls to flow to and from our softswitch without loosing call quality or volume. The solution must also mutate so that if the country notices our service and blocks our app or IP. Your solution will mutate or change IP so that customers will always be able to make his or her international calls without blockages All softswitch functions must work after deploy...

Need experts on freeswitch , kazoo pbx, etc

mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

...would have an access to the billing portal through and they format of displaying client billing information is subject to the requirements 3) PBX portal - This is where the real work happens, building the pbx requires your knowledge in running native base code such as Golang to create api with connecting feature to the pbx system. The pbx module would be the pbx system dialplan design using asterisk and it should generate dialplan base on the asterisk code when that front end on the self service request configuration from the pbx. 4) Payment gateway - The portal must facilitate with payment gateway and it allows clients to see payment history on a click away. The historical data consist of their subscription and call usage where they can refer to instantly and this payme...

need to make dialplan for realtime queue using mysql for asterisk system

you have to install freeswitch with astpp billing panel. from Astpp all uer & DID, trunk will manage. Voip call should work between PC to PC & PC to PSTN.

We are currently looking for a new Softswitch, especially for DID Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If n...Softswitch, especially for DID Management / Billing. Instead of reinventing the wheel, we thought we would put feelers out there to see if maybe someone has already built one that a project has been canceled, that may be of use to us. We have very specific requirements and are happy to modify a current system. If not, we are looking at creating one from scratch. We prefer if the project used Freeswitch / Kamailio instead of an...

High knowledge expert on Freeswitch telephony programmer. Deadline: 30 days

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I need tools to monitor FreeSwitch PBX

I have instslled ASTPP which comes with freeswitch I need someone to configure WebRTC clients to connect web phones For example And provide " how to " guide

We are looking for an expert who can help us to make our webrtc client working with opensips. We have Opensips as SBC and FreeSWITCH to handle media + call routing logic. If we connect our webrtc client with FreeSWITCH directly then webrtc working well but when we connect the webrtc client with opensips then outbound and inbound calls are not working. Please bid only if you have worked on similar task.

mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

ICTCore core is open source freeswitch based unified communications framework for developers and integrators to rapidly develop ICT based applications using their existing development skills Following you will find more details about ICTCore communciations framwork the Fax server software developed over ICTCore communications framework

VoIP Expert, VoIP developer for Freeswitch, Fusion PBX, to develop an API that allows integration in our application, where users should be able send and receive text messages, initiate, answer, transfer or put on hold a voice call, listen to VM's and call recordings, be able to send and receive Fax.

mid-register support / Dynamic Routing / Dialplan manipulation we require someone to review our setup... setup TLS / mid-register support / Dynamic Routing / Dialplan manipulation etc. Needs the ability to work Eastern Standard Time hours for supporting us and our clients

PBX telefon centrali uzmanı Genel Nitelikler - İleri Düzey PBX Sistemleri ve Voip Gatewayler hakkında bilgi sahibi. - İleri Düzey FusionPBX, Freeswitch, 3CX telefon centrallerinde uzman. - Sip telefonları provisioning yapabilen. - Tercihen linux scriptleri yazabilen. Atomasyon yapmaya meraklı. - Network bilgisine sahip (TCP/IP, LAN, WAN, VPN) - VoIP proje ve uygulamaları konusunda en az 3 yıl deneyimi olan - WAN uygulamaları ve cihazları, internet, firewall konularında deneyimli. - İngilizce bilen. Teknik döküman takip edebilecek düzeyde ingilizce bilgisine sahip - Evinden çalışabilecek. (Home ofis) - Çözüm odaklı olan, analitik düşünme yeteneği gelişmiş, sorumluluk bilinci yüksek, - Yoğun iş temposuna ayak uydurabi...

We need a person who works on freeswitch lua and asterisks who can work with us to maintain astpp server

We want to set up a multi-tenant cloud exchange, and call center structure based on Fusionpbx or directly on Freeswitch. We are currently doing the same work on Asteriks. We are interested in Freeswitch based builds due to management difficulty. In order to integrate the structure we have already used into my new site, we need the following developments. Our wishes. 1. When a new customer is created on our own CRM application, it will create requirements such as Domains, Gateways, Inbound Routes, Outbound Routes, Extensions on Freeswitch via API. 2. In addition, Active Calls, and Extensions states that I have mentioned below are given to us via WebRTC API. DIALING RINGING EARLY ACTIVE HELD RING_WAIT HANGUP UNHELD NULLInternal states( Busy call some info) 3. Providing A...

Hi it's a freeswitch on Debian 10 fresh installation

This requires setting up a call center with FusionPBX/FreeSwitch with the following points considered - a queue for incoming external calls (I'll update the destination_number directly in the Dialplan); - each agent can receive only 1 concurrent call from the queue; - if there's no available agent, the caller listen to a recording until there's an available agent; - the agents must be called in a defined order (first agent 1, if available, if not, agent 2, if not, agent 3, and so on); - one extension to listen to any agent's call in real-time; - recording of all the calls; - shortcut to transfer an ongoing call to another user; - shortcut to pause/resume receiving calls from the queue (but not from another user);

I need someone to configure a Freeswitch/FusionPBX server. It needs: - a queue for incoming external calls (I'll update the destination_number directly in the Dialplan); - each agent can receive only 1 concurrent call from the queue; - if there's no available agent, the caller listen to a recording until there's an available agent; - the agents must be called in a defined order (first agent 1, if available, if not, agent 2, if not, agent 3, and so on); - one extension to listen to any agent's call in real-time; - recording of all the calls; - shortcut to transfer an ongoing call to another user; - shortcut to pause/resume receiving calls from the queue (but not from another user);

I'm looking for someone that can help me troubleshoot call being blocked on freeswitch and fail2ban . Ineed someone that is really good at troubleshooting acl list etc.

...Registration servers Freeswitch as the Media Server 1. Instal Kamailio OS: Debian 11.3 Question: Does MariaDB required? I prefer postgresql 2. Install Freeswitch from master( I will do it) 3. User Registraion from postgresql database, I will provide the Database and table Eg: Customer 1 - user1 at , user2 at Eg: Customer 2 - user1 at , user2 at Eg: Customer 3 - user1 at , user2 at DNS Srv Records - Cloudflare and I will configure. 4. Customer1 can call their users only 5. Voice Mail Sent to Freeswitch Customers - Use Own Music On Hold Calls and IVR - Tenant I will provide the debian server, with freeswitch installed. Setup the

Night Shift IST Product: CallHippo. URL : CallHippo was launched in 2017. It is an intelligent VoIP (voice over Internet protocol) service provider for busi...create thought leaders in the business ecosystem Exp : 1 yr + Job Description - Responsible for SIP Development experience. -Involvement in SIP and webrtc integration. - Responsible for VOIP software development. -To work on Queue, IVR and Voicemail related applications. -Responsibility of Freeswitch installation, configuration and troubleshooting. -Deployment of multiple instances of Freeswitch using a load balancer. Requirement -2+ years of experience in FreeSWITCH or other related VoIP technologies. -Good Knowledge in PBX, SIP, RTP protocols. -Experience with VOIP Software developme...

Hello, we need to setup an ICT fax server. Centos7 is installed, along with ICT FAX and freeswitch. Need someone to configure it and successfully finish the setup.

I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want to move our SBC's from FS to OpenSIPS and am wanting to do the work on our own. We will require -- assistance with creating rules in OpenSIPS Control Panel -- assistance with loading the proper modules -- training on how to write the proper logic for routing calls We do currently have 2 running systems with OpenSIPS that is maintained by a 3rd party which has been doing great for us. All updates need to be understood that the current config must still work after completion.

we need an expert in freepbx to help us with dialplan

Hi David T., I noticed your profile and would like to offer you my project. We can discuss any details over chat.

I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want to move our SBC's from FS to OpenSIPS and am wanting to do the work on our own. We will require -- assistance with creating rules in OpenSIPS Control Panel -- assistance with loading the proper modules -- training on how to write the proper logic for routing calls We do currently have 2 running systems with OpenSIPS that is maintained by a 3rd party which has been doing great for us. All updates need to be understood that the current config must still work after completion.

what I need to edit to fix that , my freeswitch is sending ack to supplier if no reply in 60 sec call gets disconnected how do Ignore that or make it 3 min instead of 60 sec/ or how to make asterisk make fake 200 when ack is requested ( When your stupid supplier is not replying to ack )

I am looking for someone to sit down with our team and train our engineers on working with OpenSIPS... we are currently very familiar with FreeSWITCH but want to move our SBC's from FS to OpenSIPS and am wanting to do the work on our own. We will require -- assistance with creating rules in OpenSIPS Control Panel -- assistance with loading the proper modules -- training on how to write the proper logic for routing calls We do currently have 2 running systems with OpenSIPS that is maintained by a 3rd party which has been doing great for us. All updates need to be understood that the current config must still work after completion.