Hello, We need to implement CGrates with our current SIP infrastructure. The infrastructure in mind is an SBC server connected to a cluster of 2 Freeswitch. We would like to implement this as soon as possible. Thank you.
We are wanting a bespoke billing solution with Freeswitch. Possibly nibblebill or vbilling would be a good start. This solution will ultimately be a class 4/5 switch, be capable of managing DID numbers, multilevel admin/reseller login, and manage billing. Full specs will be detailed at a later stage.
ENG A plugin for VoIP Client is required. We already have the VOip Server (Freeswitch + FusionPBX). Our application is built upon @ionic/angular 5.6.3, capacitor 2.5.0 Our application needs to be deployed in iOS 14 SPA Se solicita crear plugin para proyecto en Ionic Framework (@ionic/angular 5.6.3, capacitor 2.5.0), la cual permita realizar llamadas (solo audio) por protocolo SIP, conectado a un FusionPBX. Su uso debe ser de forma similar al plugin de cordova (cordova-plugin-sip). El plugin debe funcionar en versiones android 11 y superior, ios 14 y superior
We currently use Twilio/OPENVBX but are searching for an upgrade as the system is 5-6 years old and has some bugs. We are a small office, but would like a system we can scale. Asterisk, VoiP, Freeswitch, etc.
Hello we are using a open source fax portal that is based on freeswitch technology. Faxes are coming to the mailbox phone number at the server side and we can manually download it BUT its not going to the associated email address. Seems like some script issue. We used to get it at our email address. Who can help. Must have some experience. Its a simple task for the qualified person. We may become your customer for our other similar needs. I think it uses php and angular for UI
My requirement is to integrate Web-Dialer to Freeswitch to make the calls easily. By default Freeswitch has a facility to make calls by the backend or we can configure SIP details to third party dialers in making a call. For an easy way I need this Web-Dialer to be integrated to Freeswitch. Does any one help or suggest in this.
I'm starting to use Accredible to generate the certificates for VoIP School. I want to create custom certificates and badges for the school. The current certificate and badge (I think they are really ugly) i did it myself. The badges will have the title Verified (Asterisk/FreeSwitch/SIP) professional I'm uploading the current certificate and one badge I did (as I said terrible) and the VoIP School Logo
We need a WebRTC client SDK that can be implemented in 3rd party projects. Needed functionalities: WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAG...WebRTC facing side: - register to a SIP server (kamailio/opensips) - establish a chat session using SIP MESSAGE (send and receive MESSAGE) - receive audio call - receive video call - enable/disable video during a session (during an ongoing call session re-INVITE and disable or enable video) WebRTC signaling plane: - SIP over WebSecureSocket (will connect to a sip server as Kamailio/Opensips/FreeSWITCH) WebRTC media plane - codecs: 711, opus, VP8, VP9, H.264 - DTLS/ICE/SRTP API facing side: - provide an easy and comprehensive API for quick integration into 3rd party...
We are looking for an experience company or a team capable of development and future support of a custom Kazoo PBX. Experience Requirement: -Must have built something similar and demo the solution with Kazoo -Previous development with Kazoo, Freeswitch, Kamailio -Examples of previous development of telephony systems based on these platforms -Excellent understanding of pbx telephony, dial-plans, security -Excellent understanding of hosting technologies -Excellent understanding of Database (SQL) with High Availablity (HA) & data replication/sharing, DRDB IF YOU HAVE NOT DESIGNED AND DEVELOPED A SIMILAR SOLUTION, do not apply. we require an experienced team capable of full delivery on time and within budget. The initial scope of the project will include creating the following o...
Hi We are looking for someone to Build an Auto Dailer CRM using Asterisk , Freeswitch and NodeJS. Designs and wireframe shall be provided by us along with HTML and CSS of all screens.
I need a developer who can create a WebRTC server using Janus or FreeSWITCH. We have a system that works in two ways: 1(attached graphic). Calls will come in through our DID. Each call is authenticated through our API (handled by a separate developer) and then each user is placed into separate channels after API authentication. The audio stream from the caller will be listened to and processed by our audio tools. The calling system is setup through Twilio SIP trunking. 2. When each user is in a webrtc channel, allow for the ability to have one or two users join via webrtc clients such as our mobile app, browser, etc. (another developer will handle client development). We are NOT building a conferencing app, but we need the ability to accept calls and then later perhaps have a c...
VoIP kazoo /Freeswitch LB / vs Kamailio setup (custom API) page to modify SIP invite programming
Knowledge of the complex issues of FusionPBX does not exist and requires someone with expertise to review, recommend and implement system changes. It will be an ongoing project/T&M position with the right person.
Hi Nsikak A., I saw you already have a click2call developed for a client, my platform is ASTPP 5.0 with freeswitch. I need the customer to have the option to choose to receive the call on the extension or pstn, and after answering the freeswitch sends the call to the site visitor. Whenever both incoming and outgoing calls are PSTN, they are billed in astpp. It can be in the two scenarios you already have: 1. Click to dial by users 2. Click to call Thanks
Looking for someone to setup kamailio / OpenSIPs for the following - Registration pass-through SBC for Freeswitch Servers and Remote Phones... kamailio / OpenSIPs will be hosted inside Amazon EC2 - SBC Setup for phone calls with Least Cost Routing... Freeswitch will use kamailio / OpenSIPs as the gateway and kamailio / OpenSIPs will pass the call to the appropriate ITSP - Setup kamailio / OpenSIPs with local registrations for SIP Clients to handle Video from Door Phones for Remote Clients
intento cambiar la credencial de Event Socket para asi crer un Freeswitch seguro en astpp, pero al parecer no encuentra la direccion de freeswitch, e usado la documentacion de astpp pero aun asi no logro dar con esto esta es la url vim /usr/local/freeswitch/conf/autoload_configs/
I'm looking for a developer who has a lot of experience with FreeSwitch for a long-term project to build it in our cloud environment for many different use cases, therefore it has to be general to begin with and then we will build out for specific use cases. I'm looking for someone who can respond quickly, is flexible and agile with deep experience because the project is going to be asking a lot from a development pov. Time is against us and you will be building in Oracle cloud.
Hello, we are looking for a developer that can handle a couple of projects involving Astpp, FusionPBX, and Freeswitch. If you have experience with Freeswitch and Lua, you should be able to handle the projects. More will be discussed once I get an idea of your experience and hourly rate
Hi, I have an issue with Call Deflection when a mobile calls into a DDI which is then forwarded back onto another mobile. This is getting rejected when Call Deflection is turned on. Works when off but I need it on to show the A party calling number. Need some to trace and fix my SIP Trunk parameters / settings. Using Freeswitch with the same SIP Trunk it works so it is supported by the carrier. I just don't know the 3CX system enough to debug and fix.
I am looking for someone to quickly put together a recording demo using either Asterisk or Freeswitch where the user is able to do the following: 1. Login into a basic site (will provide access to a DO server) 2. Have the option to start/stop a recording (simple button should be present for the user to click) 3. Once the recording is stopped it should be saved. 4. Ability to view and playback recordings with timestamps. We do not require any options to delete or edit recordings. 5. The user will login into the site on their PC and using a USB microphone will conduct the recording.
We are currently using a custom PBX solution. The PBX server is hosted in the Vultr Cloud and is based on Freeswitch with the frontend as FusionPBX. There are few issues which have come up which needs to be fixed urgently. The SIP trunk is being provided by Voxbeam 1) All incoming calls should ring the SIP softphones 2) Outgoing is not working properly for some regions. Need to diagnose that and resolve 3) If possible, need to configure an alternative SIP trunk in case Voxbeam is failing. The process should be automatic.
We are using FusionPBX, in which, we want to play a IVR, which will be an announcement like: hello engineer, your site id: 12345 is down since 10:25 am, please reach to the site with address, 12 pavel street delhi. the severity is, 1, and priority of the issue is, 2. The numbers are coming from a data source every time announcement is played. this is done in fusionpbx/freeswitch using LUA code; as we have learnt. We need someone to write this code and give this functionality.
Need to implement Linphone on ios, to be able to receive Incoming Call VOIP Push notification from PHP (Freeswitch --> PHP --> Linphone ios to be precise). _____________________________ I have a very specific need. As my team failed to make this work. My team was able to compile the softphone from Linphone source code. Was able to write push notification script, and able to send and receive push notifications on Softphone on ios. *** The only problem was, the incoming call UI on ios was not appearing as the app is closed in backgrounnd (Push notification received though). So the call cannot be picked up. I may need you to have specific experience to make this work. Not just able to make push noitifications on ios work. So we do not waste each other's time. _______...
Hi Guys! we need Softphone client for windows and Mac for our MT freeswitch platform (Hudusoft) the client need to include : BLF internal chat video calls Transport protocolls: UDP, TCP, TLS, tunneling Registrar Support Proxy Support Outbound Proxy Support Call Mute Call Hold Call Transfer Call Forwarding Conference Calls (with local mixer and codec conversion when necessary) Click to Talk Callback P2P calls (Phone to Phone) VoiceMail Send SMS from the softphone Redial Dialpad Find-Me Speed Dials Devices Auto-Configuration (network, audio, video) SIP re-INVITE and UPDATE support Auto-find people near me Message Waiting Indications Support Multiple incoming/outgoing calls simultaneously HD quality video calls (depending on your camera and bandwidth) Remote Webcam viewing Full-Scree...
We are looking for full time freelance developer to work for a VoIP company. Skills: -Programming: C, C ++ -Scripting: bash, perl, python, PhP -Databases: MySQL -Administration of UNIX / Linux servers-Network knowledge: IP protocol (TCP / UDP), IP address / IP routing, sockets -VoIP: SIP, Asterisk, FreeSWITCH, Kamailio-Traffic Analysis Tools: ngrep and wireshark -Redundancy, cluster, cloud concepts -Knowledge of DNS, IMAP, SMTP, LDAP, Dovecot Responsibilities; You will participate in all operational processed such as: Analysis of the current network Optimization of the network infrastructure Generation of statistics Analysis of network data Comparison of various telephony systems and stress tests Generation of large number of calls for stress tests Comparison and diagnostic of var...
Hello, We are going to be voip providers so, we need an experienced person to consult us on choosing a pbx with billing system which supports both resellers and clients and is as much full featured. Preferably based on Freeswitch but we are open to other options as well. Your role will be consulting now and perhaps help implementing further on if needed. Please provide information on related projects you have worked on before.
Looking for an embedded developer who has experience in working on the telecommunication system and VoIP modules. Need to have experience in FreeSwitch and ITU-T standard. Write your experience about these area.
Hi Marcelo B., I noticed your profile and would like to offer you my project. I need basically the same thing that was done here: https://www.freelancer.com/projects/mobile-phone/Integrate-Freeswitch-Flexisip-Linphone/details What do you think?
I have an occasionally issue with audio between Freeswitch and our PBX with PBX (Sip Wise) . CAll enter Freeswitch and then continue to PBX. This issue is that we cannot hear a greetings from our PBX and it looks more random since in most of the cases it is working fine. We are looking for expert who can analyze and fix the issue within the Freeswitch.
Install 2 ASTPP servers following this doc: ASTPP1 will handle billing, and ASTPP2 here will have a role of freeswitch only (handling calls). ASTPP2 needs to be connected to ASTPP1 DB, so that in ASTPP1 GUI we see also ASTPP2. Users from ASTPP1, must be able to call to user connected on ASTPP2. In ASTPP1 GUI, we must see the customers registered on ASTPP2 (freeswitch), there is an option for that in ASTPP. We suspect, that for interconnecting, freelancer needs to connect ASTPP2 to the DB of ASTPP1 and also create a sip profile of ASTPP2 on ASTPP1, to display the registered customers from ASTPP2 (but that is a hint only); The done job must