We are a communications company in south africa. We are looking at migrating our hosted services from asterisk to fusionpbx. We require assistance in configuration and require training so we can configure future deployments.
I need someone to secure my ASTPP billing solution based on FREESWITCH.
This task is straight forward. I need a iOS linphone app that recieve VoIP push and work in background. Only bid if you have done this before. No time waster. I will need to see evidence you have done this or we start with a proof of concept. Open to other iOS mobile sip client that can work with freeswitch and recieve push. I will provide server detail - i use fusionpbx
I want hire a C developer to convert some of our interpreted LUA scripts to compiled C or C++ modules. The current modules have close to 1000 lines. The copyright will belong to our company, but the name of the author will appear in the source code. Demonstrate previous knowledge of FreeSwitch and C/C++ development is mandatory. We will eventually build a similar module for Asterisk.
Hi Friends, I need professional mobile had experienced working with OTT APP (have features call look like messenger/Viber/Whatapp) , can make call and wakeup call anytime when receive push notification of server. We use SIP Server ( Freeswitch) to develop system. Each time have call from user A -> user B, system will push notification to user B User B install mobile app ( Android/IOS) and wakeup and receive call. We can make this flow work with some device but not all device work smooth, some device can not wakeup ( android, ios too) i think the app like that ( voip ) need some special skill and tech to implement. My requirement: please see the attact file
Hi Friends, I need professional mobile had experienced working with OTT APP (have features call look like messenger/Viber/Whatapp) , can make call and wakeup call anytime when receive push notification of server. We use SIP Server ( Freeswitch) to develop system. Each time have call from user A -> user B, system will push notification to user B User B install mobile app ( Android/IOS) and wakeup and receive call. We can make this flow work with some device but not all device work smooth, some device can not wakeup ( android, ios too) i think the app like that ( voip ) need some special skill and tech to implement. My requirement: please see the attact file
I have FusionPBX which works on Freeswitch. I have discovered a bug in the system. 1) When normal external call is made from an extension number in fusionpbx, the sip header contains the extension number and DID number. This is fine. 2) When call forwarding is activated the extension number and DID number are no longer in the sip header. This means that the call cannot be authenticated at extension level when sent to the carrier. I need this to be fixed so that the as a bear minimum the extension number is present in the sip header when call forwarding is activated. Example is attached
We have an instance of a customized FusionPBX running and want to move that to a new server. The Git on which our system is based on is this: I am looking for someone who is well versed in Linux Administration, freeswitch and FusionPBX to do the following 2 tasks: 1. setup a new installation based on this fork 2. move old server data to new server The file data of our current server is on a NFS-share which the new server can connect to. The database must be moved manually. The database is Postgres. I am looking for someone with lots of experience in troubleshooting such setups so we can move our clients to the new installation. If you have any quuestions please feel free to ask. thank you and all the best, Alex
Need to setup kazoo on proxmox bare metal server with container based installation of one instance each of kazoo, freeswitch...
We are a 4 year old telecom company and we’re looking for a PHP/Laravel full-stack web developer with experience in VoIP/SIP & Twilio or similar Telecom company and having knowledge of telecom and how telecom systems work especially FreeSWITCH & OpenSIPS. You must have the following skills to qualify for the job: PHP Laravel Javascript/JQuery SSH + Ubuntu Command line MySQL GIT - GitHub Twilio Telecom knowledge FreeSWITCH OpenSIPs AWS React These skills are optional: Facebook APIs We will offer long term work every week but only if you prove your skills in the first week or so. We are only looking for serious developers and to prove that please fill the attached document, upload it anywhere and add the link to the project proposal. Your project prop...
Hi I'm using fusionpbx 4.4 vs freeswitch 1.8.1. I having a problem with 30-50 cc but cpu haved load over 70%. Server 12core 24GB Ram.
Hi, do you have good experience with Kolmisoft M2 softswitch based on freeswitch ? we are facing wrong disconnect cause. Regards,
Need help with configuration in call handling.
build fs , with mod json cdr,mod XML curl, mod fail2ban,mod fifo
i have a astpp VoIP softswitch which is opensource. I install that yesterday. and after install I found there is some bugs in the lua scripts. I am looking for some freeswitch VoIP expert who can help me to fix those bugs. so person need to have previous experience on astpp and also freeswitch. if you have been work with astpp lua scripts then please apply. ready to start soon
Hi. I am looking for the following services 1. Setup Freeswitch on our AWS VPC so it becomes a functioning telephone system. We run a multi-tennanted SaaS service I want to connect to this as the registration point for agent desktops over WebRTC. 2. Allow programatic control from our SaaS platform for the creation of hunt groups, automated attendant menus, call routes, dial plan and agents. 3. Setup two types of calling use a) Agent browser conencts to freeswitch over WebRTC, but call is sent out on SIP to our SaaS platform for call termination b) Agent browser connects over WebRTC and call is placed directly over SIP from freeswitch to SIP trunk provider. During the active telephone call we will require an API or trigger from Agent side to force the telephone cal...
Hello, we need somone to install a webphone on an existing Debian 8 Server that has apache + SSL already installed. Use webphone found here: We have a freeswitch Server installed and ready with test extensions. We need this done today! We are willing to pay a bit more for the rush. So, Please respond to the task with "Dingo I'm Ready" at the beggining of your message so that I know you have read.
Im looking for a freeswitch live stream video developer with amazon aws experience with windows docker ec2 and verto
I have install Fusionpbx and set gateway (working), extensions, inbound and outbound rules, registered SPA508G phone. Inbound calls work, outbound calls and extension calls all give Temporarily Unavailable. I'm sure it's a quick fix for someone that knows Fusionpbx, I come from asterisk/freepbx, so it's all new for me. I will give teamviewer access to both linux box and Fusion website.
I am looking for a FreeSwitch live stream video developer with Amazon AWS experience with windows, docker, ec2 and verto
Configure and maintain a Sangoma SS7 SBC. It has SS7 connections with operators and a Sangoma Freeswitch setup.
Hi, Looking for some one to configure mod_nibblebill in freeswitch to use it for billing. regards
Looking for a professional company that can develop a reseller VoIP platform based on Freeswitch. The user interface has to be unique to us with the ability to integrate into a WordPress website too. Branded Mobile apps, webrtc, Windows and MAC, will also be required. Full feature list will be provided ahead of awarding.
Hi, I am looking for someone with Freeswitch experience to help me with this task. Scenario: - Caller 222-222-3333 calls x201 - x201 presses Transfer on their phone, and dials x202 - x202 rings, caller ID number shows x201 - x201 presses Transfer again to release the call and continue the transfer - x202 still sees "201" as Caller ID, until he picks up the call. Once he picks up the call, 222-222-3333 appears on the CID. I want to change the way the last step in the above scenario works. Once x201 presses Transfer again to release the call and continue the transfer, x202 should see 222-222-3333 on his CID display. This worked for us when we used asterisk. In , setting connected_line_method=update made this work correctly. If anyone has the right experience, I will be...
Looking for a VoIP architect, engineer and DevOps specialist. Project involves the design and implementation of a highly available & secure VoIP solution using Kamailio SIP Proxy and FreeSWITCH on a specific Linux distro. You must have implemented HA/Redundant VoIP networks capable of not dropping calls live during a server/site outage. Seeking DevOps orchestration method of provisioning changes using Gitlab and Ansible. More specific details and requirements to be provided if your bid is accepted. Opportunity for couple engineers for a few weeks. Must work as a team. Hourly only. (Se habla español).
Hi all. I am searching voip engineer who have good experience in FreeSwitch and KAZOO. There is maybe one time job or permanent. Depends on your availability.
develop a dialplan/RPC that transfers the bleg of a call to the play back of an audio file. and release the a-leg to be hung up. after playback of file to bleg, the bleg is disconnected/hungup
Nehos is an Australian Broadband Internet and Telecommunications Provider offering competitive, reliable and quality services to Business. Our main website - We have been using OpenSIP’s (single instance) as the core of our voice infrastructure for the last 10 years along with Freeswitch gateways but the current setup does not have much redundancy. We have attempted without success to upgrade our current setup to OpenSIPS 3.1 as a cluster for redundancy and load balancing. Our current setup is single public IP address using a destination NAT rule on a Mikrotik Router to route SIP traffic to a single OpenSIPs instance. All servers are running on Proxmox LXC containers. OpenSIPs is using a MariaDB cluster along with RedisDB for storing data. We need to be able to fail
I am providing sales & support to my customers who come to my website. I need to set up something with which customers can call me directly from my website and i can receive their calls via my browser or admin panel or any customized app or android phone softphone or app. clet me know how you can help configure this. 1) I am looking for a preferably free soft phone solution could be freeswitch or webrtc,zoiper or whatever as far as my basic requirement is met. 2) what will be the infrastructure needed ie. linux machine etc. 3) will they be able to directly call me from my website. Please share any demo or sample snapshots to better understand Kindly explain in as many details if possible or let know if you have any questions
Hi, I am looking for a softswitch class 4/5 based on opensips and freeswitch. Could you create this and what is your price ? Regards
I need a softphone (ish) application that can monitor a Sip extension (freeswitch/fusionpbx) display a popup showing incoming call with caller ID number and name. When the call is answered by clicking the answer option on the popup or when answered via the IP desk phone (yealink t46g) I need it to open a url in a default web browser and have the ability to append the callerID number at the end of the url. (Api integration, and the ability to open a program would be great, but not needed at this time) When the call is answered by clicking the popup I need it to control a yealink phone (t46g) send the call to the yealink phone as an answered call on speaker phone... Click to call on the computer to be able to dial to the yealink phone is also needed. What I am trying to do use to be ...
Hi Amrit M., I interested in updating our mobile app for our freeswitch based system. I'd like to include text messaging as well. Can we talk further?
need to install and configure a system that will allow termination of sms using 50 modems connected to a usb, actually i use linux with chan dongle and astersik fro voice termination, but it does not support sms termination. if it can support it it will be great. otherwise i can use a different module/platform, like freeswitch, Ozekisms, or also Diafaan configuration on windows is acceptable.
need expert to insatll g729 codec for pass calls.
Hello, need voip expert , need softswitch with 5000 CC , no reporting , no extra feature . One Customer >> My Switch >> ONe Vendor
Hi Web&Mobile App Developers, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for two developers with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Pur budget is about 500$/month each. Please ping me if interested! Thanks a lot for you time.
Hi Saif, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for a developer with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Please ping me if interested! Thanks a lot for you time.
Hi Arpit, my name is David Ortega, I act in behalf of a spanish voip company. We are looking for a developer with experience in react, node, freeswitch and docker. Experience with Elasticsearch would be nice also. It's a long term position. Please ping me if interested! Thanks a lot for you time.
I am looking for an EXPERT with opensips and astpp with multiple freeswitch to set up a complete switch in a cluster with centralize DB and billing or . When optimize the only limitation of Concurrent calls and cps would be hardware You must use the latest stable versions of everything. you must provide sample of similar work to be considered for this project. Do not waste my time if you have not done this before
Need to tune configuration for iternal domain calls .
Hie I need someone to deploy witch on my AWS server.I need someone who can configure an Outbound call center based to my requirement
We need to setup a ITSP phone number to BigBlueButton conference bridge, the purpose of this is users can join webconference meeting from their Mobile phone(call). For this we need to configure FreeSWITCH to receive incoming calls via session initiation protocol (SIP) from our nexmo(ITSP) provider. We are looking for someone who has deep knowledge in freeSwitch and BigBlueButton.
Hi Fellow VoIP experts, We are working on a leading web-conferencing open source platform for online learning. This platform uses Freeswitch with Opus codec through WebRTC; so directly through the browser. We made some tests and the audio quality is a lot better on Jitsi, we want to achieve the same quality through our platform. If you are a certified VoIP engineer or have a lot of experience configuring Freeswitch for web usage with opus codec through WebRTC we would be glad if you could help us. So your mission is to help us understand Freeswitch mod_conf opus codec and achieve a better audio quality result.
Minor mod_xml_curl to fetch dialplan and gateway config for FreeSWITCH servers. Implement Fail2Ban and UFW. Have the APIs in place. Just need some help on how to fetch them and implement them in FS.
Hi Seby Francis K., I noticed your profile and we would like to have some help in freeswitch configuration. Would you be interested?
Freeswitch installation need to be configured for round robin distribution
I need help on building a simple API for FreeSwitch. It is a small and quick project I want to implement at least 4 endpoints reloadacl reloadxml status sofia profile status A server with loopback4, node and freeswitch will be provided.
TWIN бизнес платформа для создания голосовых и чат ботов автоматизируем процесс коммуникации между компаний и клиентом
