Opensips a2billingcông việc
Chào bạn, Tôi tên là Tiến, ở tại Sài Gòn; cần tích hợp OpenSIPs và FusionPBX (FreeSWITCH). Regards,
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 am seeking an experienced developer for a complete migration of my current A2Billing system, emphasizing user management, call routing, and billing system functions. Key responsibilities: - Migrating user management and call routing functions - Transitioning billing system seamlessly, ensuring no disruptions - Developing, testing, and implementing the migrated system Ideal Skills and Experience: - Strong knowledge and experience in A2Billing migrations - Proficiency in database management and migrations - Excellent problem-solving skills - In-depth understanding of user management and billing systems - Experience in telecommunication systems There is no strict time limit for this project. However, efficiency and quality of work are paramount. Potential freelancers shou...
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...
I am looking for a developer who can open a link with caller and destination for every call via our Asterisk telephone system. Todo: We have A2billing and Freepbx (Asterisk). On each call open a URL like Example: Requirements: - Experience with Asterisk Ideal Skills: - Proficiency in Asterisk configuration and scripting If you have experience with integrating phone systems and web applications, please submit your proposal.
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 need to install a2billing + freepbx on my server. Create a SIP account and configure a trunk. It must be proven that it works. within 24 hours.
Necesito instalar freepbx + a2billing + 2 troncales en red local y 2 usuarios para llamar.
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.
I am looking for a freelancer to build the latest version of FreePBX system with A2billing. It will be preferable to have it in a docker, although not necessary. I understand that A2billing is no longer supported; therefore the candidate can install and configure any open source calling card platform that can cover most features of A2billing. Thank you.
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?
I am looking for a skilled freelancer to help me with a WebRTC setup on an A2BILLING/FREEPBX server. I have answered a few questions to best guide the freelancer with my needs: No, I need to configure a server; both inbound and outbound calls are necessary; and I already have SIP trunks. I require someone with experience with WebRTC integrations, VoIP gateways, Linux and FreePBX/A2BILLING setup and management. The server will be used for inbound and outbound calling services, and must be secure, reliable, and scalable. The ideal candidate should have good communication skills, and be able to work in a timely manner to complete the required deliverables. The candidate will also need to provide training and support in the set up and maintenance of the WebRTC system. If yo...
...seeking an expert for the installation and operation of a VoIP system using Asterisk and A2Billing. The work will involve the following: Full installation and configuration of Asterisk and FreePBX on an AWS Ubuntu server. Full installation and configuration of A2Billing for managing customer accounts, billing, and prepaid services. Configuration of GSM gateways to work with Asterisk / FreePBX. Configuration of a mobile phone application to work with the system (such as Zoiper, Linphone, or Bria). Ensuring that customers can easily download the app and set up their accounts. Ensuring that the system is secure and regularly updated. I am looking for someone with prior experience working with Asterisk and A2Billing, as well as configuring GSM gateways. You should be a...
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 in need of technical support with A2Billing, which is capable of handling thousands of calls per second. Unfortunately, I am having trouble dialing out and I think the lookup table might be corrupt. I do have access to the server log, and I know the IP address. The version of A2Billing I am using is 2.0. I am looking for someone experienced in A2Billing that can diagnose and repair the issue promptly.
I am searching for an experienced Freelancer to investigate why my web interface is not working for FreePBX and A2billing. My server is running on RHEL/CentOS, and I am not sure if the web interfaces are producing error messages. Interestingly, I am finding that other parts of the system are still functioning properly, so I need someone to debug why the two programs are not working. If this sounds like a job you're interested in, I would love to hear from you!
Installation of A2Billing latest and asterisk on a debian server. For future integration with linphone SDK I want someone who has past experience in configuring asterisk to work with A2Billing. Your job will be to and install A2Billing billing 2. install and configure Asterisk with cdr to be working on A2Billing I should be able to an account from A2Billing and register on a softphone. 2. Make calls using that account 3. CDR should be made in database table and I should be able to see it in A2Billing billing. You also need to tell me which configuration files were changed so that I can do this myself next time in case of server failure or new installation.
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.
I need something added to voipswitch billing I will explain We work with shipping companies and they need their invoices split on department codes. perhaps you have a solution for this. at the moment we are using a very old a2billing platform and in the customer set up we have a tick box and when we tick this box the system will do the following. the customer comes in with dial string +44203123456 and because the tick box has been activated the system will activate an IVR and ask for a department code followed by the # key. customer can now enter a random number between 3 and 9 digits followed by # key and the switch will finalize the call. In de CDRS we have an extra field where the departmental code is entered and now at the end of each month, we can group all department codes...
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 & ...
I need help about A2billing. I have a2billing customize switch server one freelancer person customize this and now he is not responding . i have some issues or bugs. Lists are give below 1. I have installed a call filter module there some issue. 2. Ip to ip call is not working 3. Alphanumeric caller id is not passing 4. other samill issues also occuring
I need help about A2billing. I have a2billing customize switch server one freelancer person customize this and now he is not responding . i have some issues or bugs. Lists are give below 1. I have installed a call filter module there some issue. 2. Ip to ip call is not working 3. Alphanumeric caller id is not passing 4. other samill issues also occuring
...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
