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    2,000 a2billing softswitch jobs found, pricing in GBP

    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...

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    I am seeking a proficient individual with experience in VoIP softswitch set up. Key Responsibilities: • Manage inbound and outbound calls • Efficiently handle call billing • Develop and integrate advanced routing capabilities Ideal Skills and Experience: • Proven experience in establishing VoIP softswitch • Familiarity with managing both inbound and outbound calls • Experience with billing processes connected to VoIP calls • Knowledge in advanced routing techniques is certainly beneficial however specific routing requirements can be determined at a later point. Additionally, specifics for call billing features will be decided in later stages, your insight would be appreciated. If you have the relevant experience and are up for the chal...

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    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.

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    ...seconds): Start with a dynamic, high-quality logo reveal of SipPulse to establish brand identity. Background audio: An upbeat, tech-inspired soundtrack begins. 1 - Company Introduction (5-10 seconds): The texts are in portuguese: Company Name: SipPulse Slogan: Software que conecta as pessoas Picture: The company in the annual meeting 2 - Service Highlights (10-20 seconds): Key software: SoftSwitch Session Border Controllers Multi-tenant PBX Multi-tenant Contact Center Interactive Voice Response Classificador de Chamadas Discador Preditivo Business Support System Short clips of developers working, team discussions emphasizing teamwork and expertise (stock) Display brief, animated infographics illustrating key services: custom software development, telecom solutions, and ...

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    We're in need of an Asterisk PBX expert capable of receiving text confirmation codes via SMS and converting them into multi-language voice IVR calls. Key Tasks: - Set up Asterisk system to receive SMS confirmation codes. - Convert the texts into voice IVR calls. - Ensure the system supports multiple languages/Libraries. - Forward calls to SIP trunk to a voice softswitch Ideal Skills and Experience: - Extensive experience with Asterisk PBX systems - Experience connecting Asterisk to SMS system to recieve the Text - Expertise in SMS/Text to voice conversion - Multi-language Library support capability - Background in telecommunication systems. In order to complete this project successfully, you should have a strong command of the Asterisk PBX system and solid experience i...

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    Currently I am using VOS3000 Latest Version, I need to be taught how to use it. My client will connect to the VOS3000 using IP and register. and the VOS3000 will set the Route for each client to the Operator's SIP CID. either use one CID or Random CID (Roundrobin, multiple SIP Operator Destination) and set each DID from the SIP Operator to each Client. and how to use a gateway that connects to several SIP providers. and how to use other features on vos3000. a brief overview as in this image.

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    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.

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    Necesito instalar freepbx + a2billing + 2 troncales en red local y 2 usuarios para llamar.

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    I am looking for an expert freelancer to help me set up a VoIP switch for my project. I am open to suggestions and do not have any specific requirements. Skills and Experience: - Expert level knowledge in setting up VoIP switches - Strong understanding of softswitch technology - Experience in configuring and customizing VoIP switches - Knowledge of SIP protocols and codecs - Familiarity with network routing and troubleshooting - Ability to integrate the softswitch with existing systems and infrastructure

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    Hey, I need a voip expert to install sippy softswitch on my server.

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    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.

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    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...

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    ...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...

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    I use a fusionpbx softswitch and I want help in overriding or replacing the hangup cause from 480 to 503 and I couldn't do it, pls bid only if you're a fusionpbx or a freeswitch expert

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    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.

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    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!

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    I am looking for an expert softswitch administrator who can set up a softswitch for me. The main purpose of the softswitch is routing and management. I need someone who can act as both my admin and goto guru. Experience: - Ideal candidate should be an expert in softswitch administration. Preferred Platform: - I am open to any open-source softswitch software as long as it is the best available. Skills: - Strong knowledge of routing and management - Experience in softswitch administration - Ability to troubleshoot softswitch related issues - Excellent communication and collaboration skills If you have the skills and experience I'm looking for, please submit your proposal. I'm excited to hear from you!

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    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.

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    We are a VoIP company looking for an experienced app developer to create a VoIP dialer app for Magnus Billing Softswitch. The app should have the following features: -Easy onboarding -Signup/sign-in wizard via SMS and email -Wallet -Top-up wallet with credit card -Upload to app store -Active/deactivate plans We need the app to be developed for both Android and iOS platforms and to integrate seamlessly with Magnus Billing Softswitch. The app should have an intuitive, user-friendly interface and be secure and reliable. If you are interested in this project, please provide a proposal outlining your approach, cost, and timeline for development. Please include examples of similar projects you have completed, as well as any relevant experience in developing VoIP dialer apps. We...

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    Must have strong knowledge about VoIP, PBX Systems, Toll Free Numbers, SOMOS, FCC Registration, Regulatory Information, Guidelines, Filings, Interconnect, A2 Billing, SoftSwitch etc We are seeking a consultant/advisor with knowledge and experience in the Telephony Service Providers industry to provide advice and consultation on business plans. The consultant/advisor should be based in the United States and familiar with all aspects of running a business in this field. In particular, I am looking for help with identifying and analyzing markets, developing pricing models, and creating marketing strategies to keep us competitive and profitable. The consultant/advisor should also possess technical expertise in VoIP networks, voice quality, and emerging trends in the industry. The succe...

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    The main part of a project is a creation of software which will help to manage GoIP equipment (Hybertone manufacturer - gateways and sim banks). It should be probably based on a softswitch (Asterisk is NOT OK here). All the features which will be included should be available in the interface - Some basic features are: 1:Simulation of Human behavior -Generation of the flow of incoming calls -The list of 'preferred' numbers. -Imitation of SIM card movement around the city. -Generation of the flow of incoming SMS and USSD requests. -Implementation of daily and weekly cycles of human activity. -Time slots for activation and time slots between calls. 2: SIM cards Operations -The IMEI substitution using an actual base of devices. -Black lists and white lists of phone numbers t...

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    I need an sip voip application that runs on Android phone (4.0). the app act as sip voip gateway termination device. It can receive calls from sip server (registerd with sip softswitch) and forward it to gsm network. The application should then forward the audio and convert from SIP to GSM and vice versa.

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    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...

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    ... We are a VoIP Solutions Provider that is currently looking to move away from our FreePBX systems we host for clients to a Class 5 Softswitch/PBX. I have compiled a list of features we want but don’t currently have with FreePBX and then the top features of FreePBX that are an absolute must have in this development. This would need to be a linux based platform capable on running on multiple dedicated servers with iSCSI storage. Ideally, we would like the platform built on a AlmaLinux or Rocky Linux. We would want the ability to have this system deployed on multiple server stacks in multiple datacenters for hyper availability. Also, to pair with the softswitch we would also plan on building the following in the future: - An iOS and Android app for our customers to ...

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    Hey freelancers im looking for build a android mobile dialer for making calls abroad it will be used ans connected eith our softswitch server where will be the all routes rates and routing data

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    Hey i need a complete VoIP solution + IPRN SWITCH

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    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

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    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

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    ...Freeswitch Administration, LUA programming, Opensips/Kamailio script Programming and good exposure to VoIP Gateways / Servers / Computer Telephony Integration, using signaling protocols, SIP, and media protocols, IVR Programming, and Customization in the Fusionpbx in the Dialplan , Core modules and AGI/AMI programming using C/Python /Perl/Shell. · Working experience with products like FreePBX, A2billing, FusionPBX, OPenIMScore, and Vicidialer · Design and Development of VOIP solutions, VOIP Support, Deployment, Integration, implementation, Bug fixing, dial plan programming, Support, Testing, and customizations of products, also understanding customer requirements and implementing solutions, proposal preparation and negotiations with the client, dealing with technica...

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    install a2billing on my server

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    We need a Softswitch developer

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    Create an api in asterisks softswitch to handle calling on android and IOS linphone apps The call will work via Local number that is instead of the app calling via internet VoIP. , it will initiate the call via the app and the app fetches the Local number from the astericks softswitch. So when caller hits the call button the api fetches the DID local number from asteriks softswitch then places the call to the destination this works silently and there will be no tap tap sound . And if there is no internet in the phone it will send the call via thesame Local DID number in the astercks softswitch like this Country code,, destination number,,# that is in a pinless calling manner. But this way will make a tap tap sound . There are many apps that do ...

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    I have recently taken this job off a developer that has been slow in completing the development. Therefore I need 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 candida...

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    looking for VoIP/SMS tech to program and maintain Softswitch to allow clients access to our routes.

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    Dear All, We have our PBX System, Forwarding portal, SoftSwitch, and Custom GUI for VICIDIAL available at your service. Now you can buy your white-labeled above-mentioned software by directly contacting the developer (iKcon Infotech). No Limit on SoftSwitch and PBX. Highly customizable. Easy to use. Best in the class Forwarding Portal. Interested Clients, Please contact me to know more details. Looking forward to getting a response from you soon. Warm Regards Shakti Rawat

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    Hi, Eremin. We wish install Kamailio on our local Data Center with SSL/TSL Encryption and SIP Redirect (customers will register using Kamailio and Kamailio will protect our Softswitch Class 5 redirecting these SIP Registers (SIP URI) to our private cloud. Kamailio will hidding our Softswitch Class 5 from public network to protect our infraestructure from DDoS and invasors. Also, we need a Dashboard/Web Panel to create SIPTrunk profiles on Kamailio for our customers and partners.

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    We already have a Voipswitch developed on Astericks and also FreeSwitch and we also have IOS and Android a...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 deploying your voip antiblock solution. Note that you need to know how to achieve this solutio...

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    The istallation of the softswitch will be used to set up a VoIP platform for a voip retail business start. The softswitch will be installed on a cloud server, and will be configured ready to start using and intergration with an app. The last step will be testing it with an app.

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    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 not, we are looking at creating one from scratch. We prefer if the project used Freeswitch / Kamailio instead of an asterisk.

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    Hi Arpit M., I have a MAgnusBilling softswitch based on asterisk and I have some issues that I want to fix. Can you help me for this ?

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    Looking for someone to install OV500 Openvoips softswitch on our VPS Server.

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    May 03 15:25:56 systemd[1]: Starting The Apache HTTP Server... May 03 15:25:56 httpd[32075]: (98)Address already in use: AH00072: make_sock: could not bind to address 0.0.0.0:80 May 03 15:25:56 httpd[32075]: no listening sockets available, shutting down May 03 15:25:56 httpd[32075]: AH00015: Unable to open logs May 03 15:25:56 systemd[1]: : main process exited, code=exited, status=1/FAILURE May 03 15:25:56 kill[32079]: kill: cannot find process "" May 03 15:25:56 systemd[1]: : control process exited, code=exited status=1 May 03 15:25:56 systemd[1]: Failed to start The Apache HTTP Server. May 03 15:25:56 systemd[1]: Unit entered failed state. May 03 15:25:56 systemd[1]: failed.

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    we need help in configuration of asterisk and and a2billing

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    Input (provided by us): - Yeti switch installed on a cloud server ...assigned, meaning all calls made from customer with prefix 100001 should be routed towards VOIP account 1). - Set ratings and tariffs to 10 destinations for each VOIP account (tariffs will be provided). - Configure customer in Yeti switch. - Provide documentation on all configurations done. - Have 1 hour Q&A session. Test scenario: We do a live test where we act as a customer (please advise softswitch capable of doing IP authenticated SIP calls) and make a call to our Yeti switch. We dial destination number with one of the prefixes created and call is routed through VOIP account mapped in Yeti to that prefix. Destination number receives a call. We check and verify account selection, rating and pricing is done...

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    We need to configure our softswitch to be able to perform functions below (Yeti supports all functions listed). Yeti is already installed on a cloud and all we need is someone with deep VOIP knowledge to assist us in configuring it through the webUI. • Platform should have ability to store different VOIP accounts and make calls using this VOIP accounts. • It should be possible to set rating and pricing for each configured VOIP account. • Platform should have ability to accept SIP calls from external party and redirect it to dialed number (destination). • Platform should have ability to configure prefixes/codes for each configured VOIP account. When there is a SIP request for a call, dialed number (destination) will contain this prefix before MSISDN of the di...

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    Looking for A2billing installation on Centos 8 with latest stable Asterisk Version. Features : - 1) PC - PC 2) PC- PSTN Calls 3) Callback 4) Inbound DID 5) A2billing integration with freepbx 6) A2billing integration with Vtiger

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    A2billing installation on centos 8 with latest Stable asterisk. Feature : PC-PC Calls PC-PSTN Calls ANI Callback Web Calls Back Inbound DID A2billing integration with Vtiger CRM

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    We already have a Voipswitch developed on Astericks and also FreeSwitch and we also have IOS and Android apps t...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 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...

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    I'm looking for a technical person who has good telecom experience and can help us developing ringless voicemail feature. There are very few companies are doing this and we have no idea how do they do that. Here is detail about one company that prov...Desire Skills Freeswitch Skills: FreeSwitch, VoIP See more: open source technologies pvt. ltd, open source technologies , ringless voicemail, professional voicemail recording, audio voicemail greetings, british voicemail recording, recorded voicemail greetings british, voicemail voice, voicemail email elastix, openser voicemail asterisk, voicemail greetings, integrate voicemail a2billing, a2billing voicemail, multiple telephone numbers voicemail, open source ajax stock charts, open source bulk sms, marketing open flash...

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