A2billing opensipstrabajos

On the client side, we have configured ICE/STUN/TURN. On the server side, we have OPENSIPS and TURN. However under some network conditions, two phones can't setup the connection. Phone B can't receive the call from Phone A. We want to find someone who are familiar with opensips, SIP trace, SDP, debug etc to fix the issue for us.

i need some one install the a2billing remotely in our computer and train us how to use, like create sip, trunk, configuration the wss.

We are trying to use push notification in opensips. We used new module which provide push notification, but I think there should be other way to wake up the SIP client app. Now our App can send the token to opensips database. When the client app tries to call to another SIP client app, opensips will check that any device token is exist in database with that particular extension and if present opensips will send voip push token to iOS/Android app to wakeup and app will send re register. We have setup Firebase messaging APIs.

Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack sinc...of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem happens when the person who is using the Webphone has intermittence of internet, generating multiple connections in closing state that are observed in the Kernel log. We think that to solve this is to use an Opensips such as Webrtc Gateway and that the flooding is controlled from this point. We need a professional to help us install and configure an Opensips as a mid-registar WSS, which allows us to log the extensions found in Asterisk of the SIP type. Observation: The asterisk actualy is not Real Time, and for local devolpment with ca...

We need to develop a payment gateway in for A2billing to process online credit card payments using our Stripe merchant account. The information from the monthly invoices should be sent to the merchant each month for processing.

Saat ini kami telah menjalankan bisnis voip menggunakan freepbx dan a2billing sebagai program billingnya dan berjalan dengan baik dan lancar. Ada beberapa penambahan script yang kami kehendaki yaitu : Status pembayaran pada kolom riwayat panggilan (a2billing-customer) Filter jam pada menu riwayat panggilan (a2billing - customer) dan jumlah penjualannya Penambahan status billing terbayar dan belum terbayar pada menu a2billing-customer Pembuatan sistem voucher (a2billing - agent) yang akan mengurangi deposit agent dan bisa dipergunakan telepon Penambahan jumlah deposit client pada a2billing-agent Diharapkan yang telah berpengalaman di bidang voip terutama freepbx dan a2billing

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Fix audio issue on Asterisk. Some calls have no audio.

...net 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 over from one server to another for maintenance or emergency while keeping user location (SIP Registration) and SIP dialog information in sync...

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 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

Dear Freelance, We are looking for Asterisk and Laravel engineer to help us build a system of billing. This system will provide billing to end user. It is a customized setup and it will retrieve data from AsteriskCDR, A2billing database and FREEPBX database to create a end user billing platform.

I am looking for an EXPERT with opensips to build me a carrier grade platform for wholesale and retail. I can work with cdr without rates for now. Platform must be able to handle thousands of concurrent calls and also doing rtp proxy. Platform must have a did management feature and the ability to provisioning and turn up customers immediately, retail with1 number or wholessle with thousands of numbers. Cdr must be very detailed with sip trace available with the click of a mouse. Basically all the features of the latest version of opensips EXPERTS ONLY. If you have a customized wholesale version of a2b I am willing to look at it also. Maybe doing signaling only with a opensips proxy in front of it

i need a sysadmin to Install A2Billing + Asterisk + FreePBX

Hello i want an expert to install FreePBX + Asterisk and A2Billing.

TWIN бизнес платформа для создания голосовых и чат ботов автоматизируем процесс коммуникации между компаний и клиентом

Hi We are looking at using ASTPP 4.0.1 for billing and OpenSIPS 3.1 as a SBC. The ASTPP needs to be integrated with OpenSIPS. We are looking for a professional who can install and test this on a production traffic. please contact me if you are experienced with the above. Thanks

Configuratiion Push Server to wake up our mobile SIP dialer to a pBX

Hello, We are a very small telecom startup and need an experienced developer to set up and install ASTPP, FreeSWITCH & OpenSIps and set them all up together, link the DB's and add our telecom carriers and get everything working 100%. If you do good work we will consider hiring you for ongoing work to maintain, run and modify it for us. This all has to be installed on AWS so you must also have knowledge of AWS and must have excellent knowledge of ASTPP, FreeSWITCH & OpenSIPs and Voip Telecom. Thank you!

Hello, I need a telecom engineer or developer to set up our carriers for ASTPP. (we already have ASTPP installed with FreeWITCH & OpenSIPs). We use FlowRoute for the DID's and incoming call origination and DIDWW then for outgoing call termination we use Intrado/West.

Hello, We are a small telecom startup looking to hire a long term developer for ASTPP, Freeswitch & opensips. Must know PHP, Codeigniter, MySQL. to Start it will be around 20 hours a week but if your really good we can move you up to more hours per week. Please send your reume with experience.

Hi, I need you to configure opensips on my ubuntu server, it has to support calls from linphone to linphone, sip messages and new users creation.

Hi, I need to configure linphone to work with opensips. The phone calls works only if in the same LAN. I guess I need nat_traversal configured, but I can not figure out how to do it. Have you done it before ? Also linphone seems to have an issue with file sharing when using a custom host. Can you do it ? Regards, Dmitry

Looking for someone to create an OPENSIPS switch on the Amazon Cloud Platform. It has to be able to integrate with Jerasoft billing and Dynamic routing. The switch will totally be a "dumb" switch and only needs to redirect and terminate calls. It need not be anything fancy. To see what documentation there is readhttp://docs.jerasoft.net/display/IM/JeraSoft+VCS+Common Desired Skills OpenSIPS

Using OpenSIPS version 3.1 I would like a configuration file with the following details Mid-Registrar for multiple domains/realms (in-memory) to Primary Freeswitch Server (UDP/TLS) with Failover to Secondary Freeswitch Server Calls from carrier proxied to Freeswitch Server with Failover to Secondary Freeswitch Server Calls from UA are proxied to Freeswitch Server with Failover to Secondary Freeswitch Server Presence should also be proxied. MWI and Park presence need to be handled. NAT must be handled. I would like to set an alias tag to any contact header proxied by mid-register in the form, ;alias={contact_public_ip}~{port} I'd like to avoid using a database if possible. Not a deal-breaker, but I would prefer to run in memory. Project completed with a working Op...

We are a telecom startup and we’re looking for a PHP full-stack web developers with experience in Twilio 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 These skills are optional: Facebook APIs React 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 (we uploaded the same document in 3 file types, use the one you prefer), upload it anywhere and add the link to the project proposal. Your p...

We have opensips running on our cento7 server, we need someone who can install cgrates & cgadmin and integrate it with opensips for real time billing. your bidding price will be deemed as final price.

Hi all, I am looking for a system admin to install A2Billing on my CentOS 7 server. I have tried installing it using the following guide but it did not work: I am receiving the following error when I make a call from a DID number: pbx.c:4458 __ast_pbx_run: Channel 'SIP/DEFAULT-TRUNK' sent to invalid extension but no invalid handler: context,exten,priority=a2billing-did,s,1 I have tried this installation guide and some others on 2 different servers: one with Asterisk 16 and FreePBX 15, and the other with Asterisk 13 and FreePBX 14. I'm looking for an engineer with A2Billing experience so that to work on this task of installing A2Billing. It will be for the engineer to choose the server to use from the two I have mentioned above

Добрый день Виктор , нас интересуют работы по a2billing , а именно чтоб несколько человек могли применять один и тот же CallerID это возможно ?

Hi, We are looking for a consultant to assist in setting up a Kamailio SBC to integrate Teams & Asterisk. Brief overview of requirements: - Centos 8 - Kamailio - Support for multiple MS Teams instances - Support for multiple Asterisk servers - Certain MS Teams instances need to be linked to specific Asterisk servers Required: previous experience integrating Kamailio/opensips with Teams This job is urgent. We need quotes today to be signed off tomorrow (Thusday) and installed Friday. Thanks, Carel

Hi, I am looking for a softswitch class4/5 carrier grade, based on freeswitch, opensips or kamailio, with GUI and billing.

Hi We are using A2Billing for Calling card using access number and DID call forwarding. Suddenly the DID call forwarding and mobile app call not working - No ring tone sound on A leg side. When B party picks call the call disconnects. Have you ever fixed a problem like this? No time wasters please. 2 hours maximum - 30 USD

Hello, 1. I have an OpenSIPS configuration. 2. When I get connect OpenSIPS with PostgreSQL that time accrued error in the system: 3. OpenSIPS version: 3.1 Master. 4. PostgreSQL version: 11. 5. Someone who can resolve this OpenSIPS with this error. Thanks.

I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY.

Ola Felipe! Que tal? I would like to finish a job with OpenSIPS (SBC between Asterisk and MS TEAMS). Do you know OpenSIPS? Are you interested?

Hi Aqs Y., I would like to discuss if you could help us. I am seting up an OpenSIPs as a SBC between TEAMS and Freepbx. We are in the middle of the job. Could we hire you? Thanks! Denilson

I am looking for an EXPERT at a2billing to set up a complete switch for wholesale and retail with self service portal. All features must be configured and works perfectly. When optimize the only limitation of Concurrent calls and cps would be hardware You must use the latest stable versions of ubuntu, asterisk and a2b.. you must provide sample of similar work to be considered for this project.

I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY. I am also thinking of using a2billing for billing.

Hi, I need to get Class 4 switching setup and am taking bids. We currently use FreeSWITCH & OpenSIPs with a cusotm BPX designed by a telecom engineer. I need someone to tell me the best open source software, why it's the best and how long it will take then to do the complete instalation.

We require someone experienced in VOIP, specifically OpenSIPS, MediaProxy, Kamailio, Billing, GUI and Freeswitch to aid in the implementation of a class 5 switch solution and class 4. Proposed design; * Two OpenSIPS proxies - All customer termination and origination from / to customer via proxies * Two MediaProxy media servers - All customer RTP is between MediaProxy and Freeswitch * Two Kamailio SBC's - Acting as PSTN gateways between OpenSIPS and carriers / TDM gateways * OpenSIPS Location / Redirect Server * FreeRadius Server - AAA * Two PBX FreeSwitch * GUI WEB: - Multi Tenant, - Multi Language (Portuguese Brazil, English and Spanish) - Resseles, - Home - Menu - Live System Reports - Rates Management - Routing Managemnt - User Manage...

I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY.

I am looking for an EXPERT with opensips to build me a carrier grade platform with all the bells and whistles .I also need to an online platform where wholesale and retail customers can purchase DiD and have it automatically provisioned so they can use it on their mobile apps to send and receive calls based on prices on out rate deck. Wholesale customers seeking termination services must be able prepay and send us traffic for termination.. EXPERTS ONLY. VOIP opensips CGrates, webrtc asterisk

Hi, We are stuck and need urgent help in getting OpenSIPs 3.0.2 working with TLS/WS/WSS listeners. Version: opensips 3.0.2 (x86_64/linux) It's getting stuck on OpenSIPs startup when using LetsEncrypt certificate. Thank you! :)

We want a Based-Asterisk system as: -5 routing devisions: Premium, Gold, Standard, Dialer, Special, -I have to upload as max as possible of rates, from carriers, for each devision, and not set the rate for client, but to set a percentage of revenue from the carrier rates, like if in Premium,...account at that time -Informations needed for signup or manual adding client: Full Name, Company Name, Address including country, Phone number, Email (with verification), accepting conditions, their IPs -Generate invoice, for client and for admin -Reports pages: CDRs, stats, income, spendings, etc. per date, per client, per carrier, per prefix called, per callee search ***PLEASE NO ASTPP OR A2BILLING OR ANY OTHER READY MADE SOLUTION. Pls check screenshots for examples we need -Quick install ...

Make software with A2Billing, or similar open source software, using custom made invoice Templates, to be filled out on our server. The Templates are made with LibreOffice. Server and PBX (Asterisk) are on different machines and networks, with different IP addresses ! All is Linux - no MS ! You must be very familiar with Billing software and Asterisk,

...existing customers in beta and ready to launch soon) and we’re looking for a PHP full stack web developers with experience in twilio and having some knowledge of telecom, not 100% required but would also be great if you have knowledge with FreeSWITCH & OpenSIPs (both open-source telecom software & PBX / SIP Server). You must have the following skills to qualify for the job: PHP Laravel  Javascript/JQuery SSH + Ubuntu Command line Ubuntu/Linux/SSH MySQL GIT Twilio These skills are optional: AWS Facebook APIs React Telecom knowledge FreeSWITCH OpenSIPs We will offer long term work every week but only if you prove your skills in the first week. We are only looking for serious developers and to prove that please fill the attached document, upload it anywhe...

I need integration of opensip over asterisk. NOTE :- DONOT BID UNTIL YOU DONOT KNOW ABOUT OPENSIPS

Hello, we need the a2billing user interface css customized and responsive. if you have idea about a2billing then its great otherwise it shouldnt be any problem for an expert in css editing. I can give you an example of what we need in pm. Thank you.

I have installed openSIPS server but need to configure it so that it does the necessary basic functions of SIP server with modules such as the SMPP for sms activated ( & ).

Chào bạn, Tôi tên là Tiến, ở tại Sài Gòn; cần tích hợp OpenSIPs và FusionPBX (FreeSWITCH). Regards,