Nagios asterisk project työt
I need to return variables between perl to asterisk pbx ivr. Also parse an xml
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi Hope you are doing well I have seen your profile seems like you are a asterisk devloper i'm also new at this world So what I need is that : I have a custom crm devlopped by Symfony 2 I have a page that show some clients with their number recently I have install grandstream UCM IP PX for to make a call center i need an script to make a simple call from extension to a custom number (of client ) to integrate it in our custom crm so the agent can click on this number then the call make automatically I have create a script with native php but the problem is that agent has to confirm the call by answer then the call is made like he need to passe by incomming call to lunch the call (this for outbound call (extension to cell phone )) I have try to solve the problem usin...
Hi, I need assistance with a voip migration project, you must have prior experience with 1) Migration from Asterisk to freeswtich 3.6 or 4.0 Multitenant enviroment 2) Setting up FreeSBC / Pro SBC 3) Dialplan setup 4) SIP Trunk setup. 5) Virtualized setup on Vmware or AWS. Project must be completed in 7-10 days.
We have already configured asterisk freepbx on premise integrated with dinstar gsm 8 sim gateway. It is also integrated with vtiger crm. We want to rehost and reconfigure it on our cloud server. We also want few email and sms integrations to be done with overall security check. We need someone who can do all above.
We have already configured asterisk freepbx on premise integrated with dinstar gsm 8 sim gateway. It is also integrated with vtiger crm. We want to rehost and reconfigure it on our cloud server. We also want few email and sms integrations to be done with overall security check. We need someone who can do all above.
Hello, We have a project in Asterisk, and need urgent support,, please message me
OpenVPN server setup needed on VPS for the purpose of remote maintenance. Remote servers (Freepbx/Asterisk) start OpenVPN on bootup and stay connected for the purpose of remote maintenance. When needed, support person would connect to VPN to perform changes.
We need to create SIP compatible softphone for our asterisk server need to include: push notification transfer and conference BLF buttons if possible auto provision template
I need a call flooding script to run on Free Pbx/asterisk simple script not complicated.
-we need this IVR system for a Radio and TV station. -after hearing the menus call will get connect to an extension. -we need reports and WAV recording. -CRM
hello sir i have a projet of voip in morocco but this period i have a problem in my system based on asterisk+A2belling ;i must do a filter to block any number enter more 12 digits and less numbers called number robotic or test number
We are looking for an asterisk / full stack developer who can help us build a call center management platform platform and dialer using Asterisk. Here is the basic outline of the project. This project will require the following. Agent Control (Manage agent and customer interactions) Call Routing Queue monitoring and reporting Provide call information (referred to as call pop-up) to agents before handing over the call 500 outbound channels sip (spanability) administrator gui phone list builder dnc database and scrubbing tool statistics page wav file uploader and player for outbound email for call results .wav file data distributing leads based on availability of agents to multiple customers the completed project must be distributable. If you have built ...
I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to GSM -Convert audio from/to SIP and GSM networks -Able to run on Android 4.0 and up. - Must run on background - Must be very lightweight to run on small memory devices Sip Requirements : - Register on Sip Proxy/Gateway - Receive call authenticated by IP, user/pass or no authentication. - Make calls with or without authentication - Forward DTMF digits using RFC2833 or ...
Hello! I am looking for an Asterisk (VoIP + SMS) developer. I have a Dinstar Device with 8 GSM SIM Cards that can accept and make GSM calls. I need those calls forwarded via VoIP (preferably Asterisks) and distributed equally among Call Center Agents that are currently logged in. This device has a capability of Sending and Receiving SMS, and we need that feature too. A software was already made for me, and it is working great and is hosted in Digital Ocean! However, I need more features, such as reports, scalability, long SMS thread, among others. However, I can no longer contact the developer. He is no longer logging in to his Skype for more than a year! Thank you! ~mark
I build Vtiger server with multiple databases for each organization. The connection to the relevant database set with the login parameters. In addition, there are multiple asterisk servers each is multi-tenant providing services to multiple organizations. I need to connect the Vtiger to the asterisk (the asterisk server ip set in the vtiger relevant DB parameter). The asterisk AMI connection is as standard asterisk server except each extension as his relevant organization as well. In addition, there is a node.js server installed that already connected to all servers and extract all events.
I NEED A SOLUTION TO FIX THIS PROBLEM. " no RBT, Fas voice recording with charge "
I need assistance in setting up my FreePBX home server. I'm using Cisco WebEx Teams and want to use the call in feature using the SIP URI provided by Cisco (example sip URI: <extension>@). So I've set up an extension with the DIAL property pointing to that specific SIP URI (which means that I'm not dialing the SIP URI from an internal endpoint). The problem is that Asterisk is using UDP and looking for _sip._udp. SRV records but WebEx is only providing TLS transport. Now I don't know how I should set this up in order to force TLS for this specific external SiP URI.
I am using call center on Asterisk there are library with and When hold and resume, can not process this event. Web phone built with angularjs Please help me.
I am an experienced PHP coder. I have built 2 major projects using Twilio voice and sms features. I have another client that would like a phone system created. I would like to explore my options with Asterisk. I would like help and guidance from someone with a lot of experience in this field to help me choose the best option and set up the proper platform. If this goes well, then it may be worth it to migrate my current twilio projects to Asterisk or another platform as well. I am looking for someone who speaks English as a first language and someone who can be available working hours of EST.
Especialista em Firewall, Switches, Servidores Windows Server, para contrato de prestação de serviços Nível 1 Avançado! Tenho um prestador de serviços, que estamos prestes a encerrar o contrato e precisarei de outra empresa/profissional para assumir! É preciso ser de São Paulo, para suporte in-loco caso necessário! Especialidades necessárias: Linux Wi...Firewall, Switches, Servidores Windows Server, para contrato de prestação de serviços Nível 1 Avançado! Tenho um prestador de serviços, que estamos prestes a encerrar o contrato e precisarei de outra empresa/profissional para assumir! É preciso ser de São Paulo, para suporte in-loco caso necessário! Espe...
Buenos días, Queremos integrar Asterisk 13 (con freepbx 14 y por debajo Centos 7) con ZOHO CRM. Los usuarios trabajarían desde PCs con windows y softphone instalados. Muchas gracias saludos
I use a production voip server. In past we had customized cid superfecta module (mainly send to growl sub-module) to match caller id with remote database and...server. In past we had customized cid superfecta module (mainly send to growl sub-module) to match caller id with remote database and send us caller details via growl desktop pop-up. this is very easy for php developer After last few system modules updates, this is not working properly (we have only caller details, nor called line info), any php developer can fix this, no need to have voip or asterisk skils. plz contact me for additional info (here : desktop growl popup notifications look - 2 files which have been edited last time - budget fixed : 80 $ in one day - bid only if you re interested and ready to start withi...
Hi, I'm buying existing code or script in various theme. It has to be fast, modern, bug free. Anything can be interesting but the following preferred: Saas, CRM, Real estate, video chat, booking system, asterisk, webrtc, etc. CODE MUST BE ALREADY DEVELOPED AND YOU HAVE TO OWN THE CODE (!!) and connected website has to be presentable with demo back end. I do not care codecanyon links or "I can do it for you OR I'm good in x or y program language". Thank you for understanding - I have very bad experience with auto bids and fake promises. Cheers
We are trying to fix the one way audio issue on Freepbx
hello, i have working asterisk fresh installed server. i need to add my voip trunk into it + my dailplan need Voip expert on urgent complete this configuration
i would like to create a dial plan and the workflow will be as followed there is 1 DID when this did is dialled it will connect to an asterisk server... this will ask for the caller to enter there pin number .. if the pin number exists it will then ask for them to set a caller id ... then it will ask them to enter there destination number which they will be bridged to
...detail is an absolute must. Issues must absolutely be resolved within a give SLA's timeline and constraints, follow ups, KPI's, resolution rates, customer satisfaction and other factors do play a significant role. You will be working with Exchange, Active Directory, Windows 7, 8, 10, Windows Server (virtual and physical(HP, Dell, Lenovo)), Linux (Debian & CentOS), MS-SQL, PostgreSQL, MySQL, PBX (Asterisk, Panasonic, Fortivoice, Mitel, Northstar, FreePBX, Elastics), Firewalls & Switches (Fortigate, Cisco, Checkpoint, Sonicwall). Networking is a must as well, switching and routing (Cisco, Forti, Netgear). As menitoned previously this is a position/contract that will incorporate multiple individuals and when we've mentioned that attention to detail is a mus...
We are looking for a guy who can create SIP based GSM gateway for us by using GSM 800 and QUECTEL M66 module so that we can connect it though asterisk application.
...asterisk16 installed via packages. I want a MINIMAL working asterisk SIP configuration to accomplish these things: 1. Three softphone extensions that can call each other. Two on desktops behind internet firewalls (different networks than the server, with NAT), and one on the FreeBSD server itself. 2. A publicly available extension, like me@, that anyone can use to make a SIP call that connects through the server to a softphone, and lets the caller leave a voice mail if no answer (which would be emailed to me). 3. Everything secure: No unauthorized access. 4. Be able to switch between RTP and SRTP by changing the configuration. 5. If possible: no asterisk database connection, only flat files. If not possible, explain why. 6. Asterisk comes up and runs as cleanly as ...
...from italian to english. This project requires experience with: - Translating websites - Fluent knowledge of both english and italian In your proposal, please share a brief summary of your experience, including examples of relevant italian to english translation work. The website address is - the Hotel della Baia link leads to an external website and isn't included in the project scope. Please put the very last word of the site's homepage in your proposal as a proof that you've read the job description. Deliverable should be in Excel format, with italian on a column, and english on another. Bold character styling should be preserved, if the document format doesn't allow it, it may be written in **double asterisk** notation. Project...
we have asterisk 11(freepbx) setup and running. need vtiger install and connect to asterisk.
We are in a requirement of sending SMSBlast/Campaign by using the SIP account. We have VICIDIAL and ASTERISK EPBX. So, by using any of these SIP accounts need to send SMS without gateway.
This project is to setup a FlowRoute SIP Trunk for inbound/outbound calling and one user (for a softphone) on a freshly installed Asterisk based server (XiVO).
Expert needed to install and configure Goautodial Call center software. the platform is based on linux centos 7. Used technologies : WebRTC Asterisk Voip
Hi, I have a asterisk PBX and my "programmer" really just did a very poor job. Basically, we moved the system from 1 server to another and again when he transfered it he did a very bad job of it. 1. The system needs a2billing added (it was on the previous one) 2. The system needs to be updated tp php7 (previous version on old server was php 5) 3. The system needs all the - customized scripts-- to work with php 7 on the new server The domain name needs to pint to a specific index file (in a folder on the server) The same file the IP address points to. He renamed a few folders also (where both the previous folder and his -- renamed edited-- ) versions are on the site. This way you can see what it - should have been-- and what he has done that does not...
Installation and customization of Asterisk 1) Install asterisk on server 2) Customize Asterisk to work as per below logic --> Asterisk should be able to receive inbound calls on DID number for different providers --> When there is inbound call on asterisk , I want system to hang up inbound call and automatically dial out/Call back on caller id from which inbound call was received. --> Once asterisk dial out then it should be capable of identifying whether call was answer by Answering machine if it is answered by answering machine then system should hang up call without forwarding to agent extension , Only calls that are answer by Humans should be connected on agent extension. In simple words i need Asterisk with Callback Featu...
We need to configure a Cisco VG224 to work in ISSABEL with all 24 analog ports. Configure dial plans and to work with more than one Cisoco VG224 in the same ISSABEL
Hi, I have a working freepbx installation. That server is working since two years without any issue. The version of the installation is 14.0.13.12. Now I want to use same freepbx server for message communication. I use X-Lite as VoIP client. I want to use X-Lite for message and file transfer as well. All extensions connected to the server could communicate each other via messages using X-Lite. Regards, ALSOFT
Need an experienced tech for Vicidial support in real time provided over Skype and email. Must be available at any time 9am-5pm PST and respond within 5min during these hours. Work load is minimal and most days does not take more than 10-30 minutes combined but must be available during these hours without exception. Must be experienced with Vicidial. Must have good conversational English. Please offer monthly price.
I need A2Billing/Freepbx Software installed on my server and configured for a soft-phone, which is a third party app on my server, the objective is that I want to:- 1. I just need help setting up dial-plan to call long distance in USA/Canada and all International and Local calls. 1-a. Dial remotely into the server, 1-b. Enter a pin number, once the pin number is entered once, thereafter, set-up a caller ID authentication on subsequent calls ! 1-c. Enter the destination that I want to call 1-d. Then the soft-phone would initiate the call. 1-e. Call history 2. Configure a method to discourage Caller ID spoofing and to activate/deactivate account - prohibit Spam Robocalls 3. Enable Search feature for admin to search by email, Account Number and caller ID, Pin Number and Account Number 4. Cre...
We are MVNO project. Asterisk Deployed. Need to connect Kamalio and SS7
I need to set a signaling gateway (like kamailio/drachtio etc) that will be use as a proxy between Asterisk server and web clients. 1. Users will use webdialer based on webrtc that will connect to the proxy using username and password. The proxy will route the signaling+RTP to specific asterisk server as a standard SIP client. The webdialer will be able to initiate calls and receive calls. The webdialer will have in addition "hold/transfer/conference" buttons. 2. CallMe button that will use a subset of the webdialer. This button should be generate a token that contain the domain, username and password. This button when clicked will connect to the proxy and the proxy will dial the preconfigured number. I'm expecting to get the server installation process and c...
Descripción de empleo Garantizar la operatividad de plataformas de monitoreo y servidores críticos DNS, correo electrónico, etc. a Nivel Nacional. Investigar servicios de monitoreo y gestión de los elementos de infraestructura técnica (Nagios, Cacti, SMS, MRTG, entre otros.). Desarrollar nuevos servicios de valor agregado. Evaluar y recomendar nuevas tecnologías en el ámbito de valores agregados. Requisitos: Graduado en Telecomunicaciones. Conocimiento de Linux y Cisco. Detalles: Área: tecnología, Sistemas y telecomunicaciones. Nivel Mínimo de educación: Universitario. PuntoNet cada día innovamos y brindamos soluciones de conectividad a nivel nacional. Somos líder en telecomunicaciones, con...
Asterisk program is being hosted by our own server in a VM environment, we will provide access to VM and internal NAT. I'm looking for someone who is experienced with Asterisk PBX platform, who can assist with our internal PBX setup, provide some support when needed, make some adjustments to the setup when required, and also who can help us to support any client Asterisk instances that we manage. Currently system needs changes and upgrades to SSL, Predictive dialing, IVR, Incoming calls, and potentially GSM hardware upgrade (need to evaluate solution) with Asterisk expert. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Other task related to asterisk server that will be needed in future:...
Hello, We need help to setup Analog Voip Gateway with Asterisk, model: ~ Remote session is provided with TMATE, so be ready for it. Thanks in advance
