A2billing opensipstrabajos
I need help about A2billing. I have a2billing server in running condition few modules are added into this but there are some issue like: 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 would like to have an application made with Asterisk , Kamailio, Opensips where we can make a call campaign and report on keypress. We want to be able to make 500 or more calls simultaneously. Please contact people who have made improvements on these issues before.
I would like to have an application made with Asterisk , Kamailio, Opensips where we can make a call campaign and report on keypress. We want to be able to make 500 or more calls simultaneously. Please contact people who have made improvements on these issues before.
Install opensips server with opensips control panel. Cgrates should be added
we need to configure the A2billing that the functions of start billing and end billing will not instruction from freepbx but from the API that is connected to the end point
we need an expert A2billing and Freepbx to configure for us a dial plan that will fit our need
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
delay the buy signal on asterisk or opensips, when the call is hangup delay between 6 or 7 seconds to receive the bye
I need to escalate my ASTPP server, putting an opensips in front to send calls to more freeswitch servers. I need to handle a minumum of 150 CPS with this setup. Maybe 8 freeswitch servers or more.
I need a softphone for my a2billing platform.
I need the system to send me invoices twice a month. The first invoice will be from the 1st - 15th of the month. The second invoice will from the 16th - end of the month. The invoice should include: 1: Total minutes used for that period 2: Total number of calls 3: Balance owed
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...restrictions, the voice is so clear and there is no such problem in the call Note 2: The funny is once I try to dial the other extension of *43 for the Echo test and our voice is still shaking If I pressed any num key twice or three times, our voice becomes so clear! I think may one of the following ideas are resolving the problem: 1- Install & Configure a Cloud SIP Proxy Server like Kamailio, OpenSIPS or any other SIP Proxy Server 2- Install & Configure a Cloud Stun/Turn Server Note 3: I can provide a Cloud VPS server to use as you need to solve the problem Note 4: 1- It's up to you to resolve the problem with any perfect method 2- Awarded will be to the one have an experience only to complete this project successful (no wasting time) 3- If the project is done pe...
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I need a script to install a2billing on issabel pbx server
I have a script to install a2billing on issabel pbx server but it is not full working it need same change to work i need a full script working a2billing on issabel Create A2Billing Database. Create files and set permissions. Run sound installation script. and i have ip is installed issabel u can used it to test 5.252.162.165/ i will give u shh root password to try it
A2billing Installation on my hardware and Basic Setup Need, call billing and Call Back Working Condition Setup Budget the A2billing=10$ the Configuration (Dial Plan, Rate Card, Billing, Extention, Trunks, Callback, Calling Card)=10$ 3. Test Call Work with Above All Functions=80$
I need to configure Kamailo SBC to connect multiple Microsoft Teams account in the same SBC server. - Install Opensips - Install Opensips-cli - Install Opensips-cp - Configuration TLS certificates - Configuration RTP Proxy / RTP Engine - Routing Inbound / Outbound - Security Example MS Teams 1 <--> Kamailio SBC <--> Asterisk MS Teams 2 <--> Kamailio SBC <--> Asterisk
Hello... we are looking for opensips developer for modifications in sip header. We want to comply our opensips based switch with stir/shaken standards and need some expert who can make modifications in our present software which is based on opensips 2.2 version.
Hi, I succeed in installing two Freeswitch server (version 1.10.3). I have step by step install guide for freeswitch. Now I want implement load balancer for two Freeswitch server with Opensips server (version 2.4.x) But I don't know how to implement this architecture. I am failed in install opensips and config opensips server connect two freeswitch server with role as load balancer. Note: I wan' install and config opensips server only. I can install Freeswitch if you need. Please help me.
Currently we have several PBXs in Asterisk and there is a problem of Flooding of port 8089 Webrtc, it is not an attack since the traffic is valid and comes from IPs of our clients, this problem h...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 Kamailio 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 cannot ch...
We are looking customised multi-tenant open source call centre VOIP solution, with an android and IoS app for calling agents. We should be able to appoint resellers, white label it for resellers, we should be able to sell b2b call centre solutions to clients in various industries. It sh...Productivity Sale Graph Real-time call status Standard Call features Standard reports and other range of reports Voicemail All basic features Inbound number Black Listing Email & SMS Module Internal Chat Module Skill based Routing with Agent Ranking Sticky Agent You may use any or many open source softwares to get these features, like Asterisk Fusion Pbx, ASTPP, FreePbx, free switch telephony, webrtc, a2billing, vicidial, magnus billing etc. The system has to be robust scalable and capable to...
A2billing install and Basic Call Setup in my freepbx
I i need to add ip wholesale auth to a2billing system, and also, know how to lower the length of secret in voip settings of all customers, i want at least 5 characters password.
...size of the signup. Call travels from their FREEPBX machine to a single central A2billing system with a facility of DID management. The trunk which this SIP service is running are directly connected to the A2billing or a gateway attaching a physical SIP/ISDN cables routing back the calls to A2billing. Each Freepbx system has a customized call reporting system that supports call center reporting and call recording with a unique dashboard base system. When a subscriber signs up with an extension, they can access the webservice to see their call center reporting and access web phone. This is available in the system now. The current new plan is to add a new Kamailio system to support the FREEPBX machines >> A2billing route which is FREEPBX machines >...
I'm using OpenSIPS with SippySoft. I want someone to update the OpenSIPS to latest without loosing any functionality and show me how it's done. More work will be provided in case of success.
I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)
I want to HA-Setup of Kamailo SBC to connect multiple Microsoft Teams account. So in this Project: * Install Kamailo-Cluster * Install Opensips (with CLI and CP) * Configure TLS Certificates * Configuration RTP Proxy / RTP Engine * Setup Inbound / Outbound Routing * Setup Security best-practice Example MS Teams Tenant A <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk) MS Teams Tenant B <--> Kamailio-HA-Cluster SBC <--> PBX (SIP-Trunk)
Please let me know if you have experience in above tools. I've some little things to do. First project is. I need to mask the RTP IP of the originating site with any IP which is not present in the server. So the termination site will see a different RTP IP than the original one. More details will be discussed on chat.
looking for customizing Linphone for iPhone and android, Need "AUTO LOGIN WITHOUT USERNAME AND PASSWORD" (username and password from a2billing table), PUSH NOTIFICATION SHOULD WORK TO RECEIVE INCOMING CALLS AND FCM MESSAGES, SMS OPTION SHOULD WORK WITH ASTERISK SIP SOME OPTIONS OF LINPHONE to simplefy linphone interface. we have a sample source code and customized linphone dialer TO SPEED-UP THE PROJECT. After finish the project, we need the final source code.
Hi. I need opensips expert for hiding or modifying the RTP IP before reaching the termination site. Means I need to modify the RTP IP from the sip trunk running on Opensips as a different IP, before it reaches the termination side. So the termination site could not see the actual IP address.
I am looking for a OpenSIPS expert to update my opensips from Version 2.2.7 to Latest stable release Version 3.1.1. You need to make sure everything is working. Including the dialplan and database.
Please let me know if you have experience in above tools. I've some little things to do. First project is. I need to mask the RTP IP of the originating site with any IP which is not present in the server. So the termination site will see a different RTP IP than the original one. More details will be discussed on chat.
Hi, I need some work finished for my a2billing installation. I had other people work on it for me, they have got it going, but I just cant seem to get all that we need from them in a timely manner. The current project of also over budget and past the deadline. I need someone who is fully versed in a2billing. Someone who knows how to get the job done, and give me a working product. Please contact me for more details.
I am looking a Technician for Freepbx & A2Billing Setup with OpenVpn & Nat Access
Hello. Need help installing and configuring Opensips
Приветствую! Нужна помощь в установке и настройке opensips.
I need to install a stable version of Opensips 3 with a web interface on a server running Centos 7. It will be used as an SBC to hide the network topology, change sip headers and load balancing.
We need to build opensips based voip billing platform. In which opensips will be able to handle customer billing, LRN/LNP, Routing, LCR, Rates, customer account balance. If anyone have good knowledge on opensips please reach to us. Please note : I will pay only the amount you bid. If you change or try to negotiate after bidding it will be waste of time for both of us.
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...
O projeto consiste em um balanceador de tráfego SIP. Os destinos devem ser recuperados de uma tabela de banco de dados MySQL. Uma ação (executar uma URL ou update no banco) deverá ocorrer se algum dos destinos retornar um código específico (Ex: 602). Não será necessário verificações de segurança (acl, etc) pois o ambiente de produção será rede local. Não será necessário adicionar serviços de RTP pois os áudios serão fechados direto para o IP de destino, resultado do balanceamento. Por ser parte integrante de um projeto, não será necessário nenhum tipo de interface do usuário. Preferência por instalaç&at...
I have installed OpenSIPS 3.1 server with UAC, CP and rtpproxy on Debian 9 server with 2 network interface. (Asterisk server - OpenSIPS - Service Provider) Now I am able to establish calls and also registering to my service provider. I need someone's help to fix the rtp traffic and routing.
Current VOIP stack Setup: 2 opensips servers (clustered), 3 rtpengine servers (clustered), 2 database servers (clustered), and would like to host multiple PBXs using opensips as a with different domains. Would like to verify current setup and also get webrtc functioning correctly. I am looking for an expert in OPENSIPS and working with webrtc experience. Thank you, Convergence
Hello I have a centos8 running kamailio asterisk and a2b, the service has chat with file transfers and of course voice and video calls, I have built it with several developers and I have also android custom built clients and apis, Im looking for someone to start installing the required above explained prefferably on debian 10 but 9 would do its a cooperative work between you and me to copy my current service from centos to debian and make the clients work, we will deliver a fresh stock debian 9 or 10 server with its public ip and no firewall
Нужно установить биллинг желательно a2billing на сервер freepbx или других программ которые похожи на a2billing Надо чтобы установить лимит минуты разговора и ихний стоимость Сервер для ip telephone asterisk
We are looking for someone to build php driven crates admin and customer portal. Modular, (my)sql based, utilizing API access to cgrates, in English. Strictly wholesale/retail billing; no reseller, no calling card functionality. Interested party must be well knowledgeable in voip, voip billing, asterisk , kamailio, opensips, linux, cgrates, php. When completed this project may be released as opensource.
() in ASW () in my AWS account leaving 100% its real functionality.
just test project there are some projects more after this.
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.