Opensipsemplois

Installation de opensips sur un serveur et asterisk/a2billing sur 2 autres serveurs

j'ai besoin de installer un serveur opensips avec une proxy rtp (rtpproxy) + serveur Freeswitch le tout avec debian jessie x64 comme sur le plan: Merci

Preciso de um WebRTC Proxy para conectar nossos antigos sistemas asterisk em webphones. Ele tem que suportar conexões ipv4 e ipv6. Ou seja, precisamos de um Proxy que faça a conexão dos clientes WebRTC tanto ipv4 quanto IPv6 e entregue as chamadas na rede local. Precisamos de pessoas com experiência em kamailio, opensips, rtpengine, asterisk, freeswitch, MariaDB.

I'm looking for an OpenSIPS expert with comprehensive knowledge in call routing. currently we can able to work with opensips but have 2 issued, I need Expert opensips for support me config Opensips: 1. Config Redirect module uac_redirect - currently we have issued when A call B, and B ring 180, after that B refer call to C, call not reach to C - I'm follow opensips Docs we have to config uac_redirect but i try with no luck 2. We use media server is Freeswitch, how can i pass a variable to Freeswitch, i try add header follow format: X-Variable, sip_h_X-Variable but in luascript of FS, can not get this variable

I'm seeking a skilled developer to create a versatile VOIP softphone app, compatible with bo...Ability to send and receive SMS & MMS messages - Auto provisioning function - In-app address book Further, ensuring seamless integration with APIs is a must, as well as a functionality syncing with an address book or contacts. The ideal freelancers for this project would have solid experience in iOS and Android app development, with special emphasis on communication apps and certificate pinning. OPENSIPS knowledge + Knowledge in API integration and VOIP technology is highly essential. Your portfolio showcasing related projects will highly increase your chances of being selected. Additionally, we stopped using Bria because of audio degradation on IOS devices. We need this resolve...

I'm currently using a private VoIP provider for my Opensips configuration and need help with translating specific tech prefixes. The prefixes involved are rather simple and straightforward. My goal is to ensure efficient configuration for smoother operations. For this job, you would need: - Proficiency in Opensips and dialplan configurations - Past experience with tech prefix translations - Ability to work with private VoIP providers Your role would involve: - Analyzing the current system and identifying the prefixes - Translating these prefixes within the Opensips dialplan configuration - Verifying the seamless operation of the system post-translation If you have prior experience with similar projects and are confident in your ability to make this translation...

I'm currently using a private VoIP provider for my Opensips configuration and need help with translating specific tech prefixes. The prefixes involved are rather simple and straightforward. My goal is to ensure efficient configuration for smoother operations. For this job, you would need: - Proficiency in Opensips and dialplan configurations - Past experience with tech prefix translations - Ability to work with private VoIP providers Your role would involve: - Analyzing the current system and identifying the prefixes - Translating these prefixes within the Opensips dialplan configuration - Verifying the seamless operation of the system post-translation If you have prior experience with similar projects and are confident in your ability to make this translation...

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

Hi I'm looking for a Kamailio / Opensips platform for routing and cdrs. I'm currently using freeswitch (with no transcoding) but i think we can save lots of cpu with a 'SER derivate'. If you can help me, let's talk :)

Hi I'm looking for a Kamailio / Opensips platform for routing and cdrs. I'm currently using freeswitch (with no transcoding) but i think we can save lots of cpu with a 'SER derivate'. If you can help me, let's talk :)

Hi, I want a billing solution to work with opesips or Kamailio. If you have developed a billing system along with LCR and Routing for opensips and kamailio please bid. Thanks

Project Description: I am looking for a freelancer with advanced technical knowledge to develop an Opensips LRN query server. The purpose of this server is to provide real-time information about the location routing number (LRN) of a phone number. Requirements: - The server should be able to handle queries for routing telephone calls based on the least cost routing (LCR) principle. - The server's response time needs to be extremely fast. - The freelancer should have advanced technical knowledge in Opensips and related technologies. - Experience in developing real-time information systems and handling large volumes of queries is preferred. NOTE : Price Negotiation will not entertained after bidding.

Hi Eremin P., We worked together on an opensips with control panel server. I need to fix the sip cap to be able to pull from control panel. Can you help me with that?

Hi Eremin P., We worked together before on an opensips server with control panel. I need to fix the sip caps so they can be pulled from the control panel. Can you help me with that?

Hello, I would like to request your assistance in installing OpenSIPS on a Debian server. Below are the details and requirements for the setup: OpenSIPS server details: Public IP: Private IP: We will be receiving calls from the IP address 24.24.24.24. Any calls received from the IP address should be forwarded back to the same IP address (), but with one modification: The media IP should be changed to 10.1.1.1. It is crucial to ensure that audio works properly on both ends of the calls. Upon completion of the task, please provide a comprehensive documentation outlining the installation steps and configuration details. If you require a virtual machine for testing purposes, please let me know, and I will provide

I am looking for an experienced freelancer to help me add CGrates to an existing Opensips server for personal use. This project requires expertise in MYSQL, VOIP, Linux , as well as setup and configuration of the CG Rates software infrastructure. I need someone who understands the requirements to integrate CGrates with my existing Opensips server, and can provide me with the necessary guidance and advice to ensure a successful result. It is important that the freelancer is experienced and knowledgeable, so that they can solve any issues that may arise throughout the project. This is the link to add my requirement, it's an example maybe the developer has a different approach:

We are looking for experienced professionals to help us build a VoIP SBC system, using Kamailio/OpenSIPs preferred. The services we would like to set up include hosted PBX, call termination, and call routing. We do have an existing infrastructure which needs to be connected to, so knowledge of this process is needed. We are also interested in advanced features such as call quality monitoring, packet capture, security measurements etc, analytics etc. There is a billing system which is a sort of SBC itself, but very small and only intended to bill call termination. Also, we have a bunch of Asterisk servers spread all over the internet acting as PBXs and sort of B2BUA proxies/gateways for the rest of infrastructure. The idea of this project is to build a geo-redundant system having...

...5089 We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following: 1) The Proxy should communicate with the asterisk server on UDP SIP port 5060. 2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 3) ALL TCP SIP traffic on 5062 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGISTER ETC) 4) Listen for incoming SIP TLS traffic on 5089 on the LAN IP X.X.X.X. and proxy this SIP TLS Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 5) ALL TCP SIP traffic on 5089 should be proxied from Kamailio/Opensips to asterisk (Not just INVITE, REGIS...

Hi; I am a private IT enthusiast, who is interested in learning more about IT. I have recently come aware about Asterisk, Kamailio and Opensips. I would like to build a lab using virtual machines to see, how all of this works. What I am looking for is a kind of tutor, who could guide me through the steps to build this lab and get it working, while explaining to me tha essential things that I need to know. I am not a professional, so I do not need much details, even though sometimes I would have to get some. If anyone is interested in helping me building this lab once with Kamailio/Issabel and Kamailio/FreePBX and also using Opensips - connect it to Jitsi or MS Teams, do not hesitate to contact me and tell me, how much would it cost. Thank you.

Hi, We have OpenSIP (with WSS) with RTPEngine configured but we are not able to make audio calls working for the webrtc based client. Our flow of calls is like this: WebRTC client -> OpenSIPS -> FreeSWITCH The system is deployed on Azure. We are looking for experienced person who has done such work and quickly help us.

We need an opensips SBC to connect our PBX based on asterisk and free switch to microsoft teams

We need an opensips SBC to connect our PBX based on asterisk and free switch to microsoft teams The goal of this project is: Configure an Opensips server with Opensips Control Panel that: - Connect to asterisk PBX server / Fusion PBX - Connect to Microsoft Teams - Let users from MS teams call users on Asterisk / Fusion Extensions and external calls though this PBX - Let users from Asterisk/Fusion call users on MS Teams

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...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-E...

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 a...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-E...

Hello, 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 iss...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 expert in FreeSWITCH, WebRTC & ...

...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, Perl, Asterisk, Freeswitch, Opensips, Kamailio VOIP, SIP, IMS, NGN, ISDN, TDM, and Telecom/Network Protocols. · Very Good Knowledge of VOIP/SIP servers, Design and Development, Support, Testing, Deployment, Asterisk Programming, Asterisk Administration, FreeSWITCH Dialplan, Freeswitch Administration, LUA programming, Opensips/Kamailio script Programming and good exposure to VoIP Gateways / Server...

Freeswitch / opensips / pbx development work

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

We need assistance in setting up OpenSIPS as a B2BUA between a PBX and an SBC. The B2B will also need to manipulate the SIP Header info on egress.

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.

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.

Looking to setup MS Teams and Asterisk/Freepbx Integration for multiple clients. I understand I need an SBC of some sort, either an OpenSIPS or Kamailio server.

Need to develop Kamailio/OpenSip-based SBC for VoIP Wholesale Session 10K Calls CPS - Unlimited Self Care Portal for End b2b Customers, ( Signup/Signin/Forgot Password/KYC/IP Addition/Deletion/CDR Summary/Invoice Generation/ Reporting Stats/Create Ticket/Payment Gateway/ Multiple Trunks ) Reference :

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

Hello, We are looking Kamailio and Opensips Expert to integrate the below Kamailio and opensips module We need a return Invite as per the below URL configuration Can you please help us Thank You

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

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

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.

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.

I am skilled voip developer and i can update opensips installation for client requerements.

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.

- Route all calls between Asterisk and MsTeams, one domain/asterisk only. - Ubuntu,Kamailio or Opensip, letsencript, and powershell will be provided by us

We need a SBC installation (Kamailio or OpenSIPS, preferably Kamailio), which will have several Asterisk servers behind, that do not use realtime (and won't use) but use a database to generate config files (both extensions and sip). Said SBC must manage the authentication, but Asterisk peers should still be independent and have their own config (things like context, call-limit, accountcode, etc.). Due to not using realtime for Asterisks, registration control should be managed in the SBC's own database. The SBC will have to load balance between the asterisk, and have failover between them. It should have the basic security modules as well (e.g. secfilter with GeoIP for Kamailio). Also it should be able to control NAT and RTP as well (rtpproxy or rtpengine).

WebRTC • Experience with Strong WebRTC platforms • Experience with WebRTC (direct use and/or debugging of the WebRTC APIs) • Knowledge WebRTC server technologies. • Actively involved in all phases of our WebRT...all phases of our WebRTC product lifecycle. • Experience developing applications on a VoIP infrastructure • Integration of Web and Mobile Applications with WebRtc Technologies for Audio ,Video , Screen Sharing and Meeting • Knowledge and experience of the following protocols and platforms: SIP, WebRTC, RTP, REST APIs, Linux • Knowledge and experience with one or more of the following: Asterisk, OpenSIPS, Kamailio • Good knowledge of media codecs, formats, transports and container protocol Webrtc • Good knowledge on Video Tran...

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 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 projects

Hi, Given: SIP-Server A (IP A) SIP-Server C (IP C) OpenSIPS B (IP B) Calls can be from A->C and C>A. A->B->C C->B->A OpenSIPS 3.2 (B) in the middle should route the Calls with a RTPProxy. OpenSIPS should generate CDRs.

I have opensips on the LAN and WAN. RTPEngine is already installed.. I need to bridge the audio between the WAN and LAN. Also LAN to LAN

We are a telecom, marketing and analytics startup 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 & PBX’s. 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 of VoIP, SMS /MMS, Call center style call queues, call conference rooms FreeSWITCH OpenSIPs AWS React We will offer long term work every but only if you prove your skills in the first few weeks. We are only looking for serious developers and to prove that please fill the attached questionnaire document, upload it anywhere and add th...