Freeswitch jobs
Requiero implementar seguridad a un servidor en astpp, la seguridad a implementar son los escaneos de extensiones sip, escaneo ssh, httpd, y cualquier otra sugerencia que propongan. por favor quien no aya trabajado con esta plataforma que no me haga perder mi tiempo. I require security to implement a server in astpp, the security to implement are the sip extension scans, ssh scan, httpd, and any other suggestion that you propose Please, whoever has not worked with this platform, do not waste my time.
WebRTC Media Server with Nodejs to receive audio data and send audio back | Test it with FreeSWITCH / Asterisk
Hi, We need someone who can upgrade our FreeSWITCH and OpenSIPs to the newest stable versions on Amazon AWS. Currently we use FreeSWITCH version: 1.10.2-release-14-f7bdd3845a~64bit (-release-14-f7bdd3845a 64bit) and the newest stable release is 1.10.8 We also need OpenSIPs upgraded to the newest version 3.3.2 we currently are on: 3.0.2 (x86_64/linux) This is a live production server so it will need to be done pretty quick in a couple hours or so. If we work well together I will have many more ongoing tasks involving FreeSWITCH, OpenSIPs, our PBX and other issues, our main telecom engineer/developer was in Ukraine and we have not heard back form him in months. Thank you! Thank you!
Hi Arshad N., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details
Hi Aqs Y., I would like to offer you my project. We are using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. https://www.freelancer.com/projects/voip/FreeSWITCH-WebRTC-OpenSIPs-Expert-Needed/details
...using FreeSWITCH (I am not sure which version) along with WebRTC for our Soft-Phones (our Hard-Phones do not have this audio quality and delay issue), the Soft-Phones have audio quality issues, static, pops and crackle's at random. I have read thru google searches and see some versions of FreeSWITCH have audio issues with some versions of WebRTC. Our Soft-phones are made with React.js I need a person who knows what they are doing, we also use OpenSIPs so the codecs in OpenSIPs might not be correct but this is just a guess. Can someone solve this for me, I have a hard time getting honest developers here, it seems like everyone says they can fix it, then I waste a week with them and have to cancel and look for a new developers, please only bid if you are truly an ...
Various tasks in freeswitch. Requirements - understanding, communication, desire to progress on the subject. long-term cooperation
We want to provide our cloud switchboard solution via freeswitch. For this, I would like to discuss the project with people who have experience in Freeswitch and API development.
Connect the Web Application with API and Dynamic Data with the Freeswitch / FusionPBX System.
Hi I need a hand with freeswitch: I have a client that needs to send calls to my freeswitch but his switch doesn't have username+password authentication. He is asking me to have my freeswitch accept the calls based on his public ip address. Let me know if you are able to help but only bid if you have extensive experience with freeswitch as this is a security concern. Max 100 euros. Thank you.
...Description • Candidate should be familiar and comfortable with Freeswitch. • SIP Development experience. • Must be aware of Sip and webrtc integration. • VOIP software development. • Good Knowledge in PBX, SIP, RTP protocols. • Worked on Queue, IVR and Voicemail related applications. • Expert in Freeswitch installation, configuration and... • Competent enough to setup daily call limit and concurrent calls Requirements · Software Development experience in Freeswitch, FusionPBX, Opensips, SIP, VOIP, SDP, TDM, IMS, PSTN, Python, Perl, Linux, and Open Source Technologies. · Strong Technical, Logical and Debugging skills with innovative and result-oriented approach ·working experience in Python, Shell, Pe...
Hello Amrit, We are an Italian internet provider. At the moment we are using a custom Freeswitch cloud pbx calle Hodusoft. We need to customize Linphone to work with. Registration is OK but blf doesn't work and we would some feature like an easy provision and contact. Can we discuss about it? M
We are looking for someone who can fix certain compatibility issues with of Flutter BigBlueButton app with latest BBB server. In the long run we'll need to include additional features as well. Flutter Code: Make sure you know what FreeSwitch, WebRTC means. Without depth knowledge on them you'd get lost
We have a requirement of Limit management in freeswitch Limit for concurrent call and Max call in a day for per destination
Looking for a Class5 Soft Switch based on FreeSwitch with Android and IOS APP. *Account Management *Calling Cards *Rate Groups / Tariffs *Call Routing Strategies *Call Rates *DID Management *Product Management *Invoicing and Billing *Reports and Alerts *System Settings *Payment Gateways
Hi Eremin P., I noticed your profile and would like to offer you my VOIP project. Cloud PBX based on Opensips and FreeSWITCH. This would be a long term project. We can discuss any details over chat.
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 ...
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...
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
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...
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
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.
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.
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...
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
