Asterisk pbx project trabalhos

Hi, PLEASE READ CAREFULLY THE WHOLE ...CAREFULLY THE WHOLE OF THIS BEFORE BIDDING We have a project for an Asterisk IP Pabx expert to configure a hosted phone system in the cloud. We require up to 10,000 extensions eventually. These extensions are to provide emergency dial out facilities for an emergency call system for elderly people. Each elderly persons home will have a unit that looks like an IP extension off the PABX. When an emergency call is made (by pressing a Help button), the IP Pabx will make a call and allow the relative or help desk to talk to the person who needs help. We want to initially set up an Asterisk system to test the service, then develop a more sophisticated service. YOU MUST HAVE ASTERISK EXPERIENCE Please make the title of your b...

...not have tariff need tariff in this with all type of customers. 2. Kazoo does not have Device provisioning in it for SIP devices as per SIP brands to program certain keys. Modules Required, 1. Retail Client Client for selling DIDs & Termination 3.PBX routing like VOIP Switch,asiign as many extension numbers to one extension,ver cool feature. to select Different trunks for different PBXs to select number of registrations per account. option must be in Retail & PBX. user interface for PBX admin & Client & Retail client. 8. Live view of SIP call with information of error. 9. SMS service like SIP simple 10. SMS delivery over DID both ways, inbound SMS & outbound SMS. Please contact only developers with the past experince & who have ...

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 in Network topology

Necesitamos desarrol...datos sean correctos, y realizar el checkout contra MercadoPago. -Una vez obtenida respuesta, reproducir respuesta (exitosa o no exitosa) y enviar resultados a otra función PHP para actualizar status de pedido -Se contempla la posibilidad de integrar audios personalizados,pero queremos nos ofrezcan la posibilidad de configuración y presentación de incluir voz automática IVR -Desarrollo sobre Asterisk ademas de los descripto necesitamos contar con, 3 sesiones de soporte adicionales de 1 hr posterior a la entrega del proyecto Tiempo máximo de entrega del proyecto 10 días Sesión de pruebas unitarias remota Presentación de propuesta de voz automática Manual y documentación de solución ...

Admin interface: -Creating carriers: Carrier name, Carrier IPs:Ports (To be allowed where calls to from), Personal Notes. -Adding numbers with CSV File:...incoming calls, for expl: a destination which we get paid $0.18 for each minute, we payout 0.17 for each minute the client makes There will be difference between Carrier Payout and Client Payout, which will be used for calculating profit, In the stats pages -It should allow high capacity, and secure -At the end we need a quick install script for all of it ".sh" including Asterisk/FS Skills: PHP, Software Architecture, Asterisk PBX, MySQL, JavaScript See more: making money with international premium rate numbers, iprn telecom, iprn providers, international premium rate numbers providers, international...

...understanding of integration with Ribbon, Audiocodes & other IP PBX devices • Responsible for planning, designing, installation and configuration of Microsoft Lync server 2013/Skype for Business and Enterprise Voice. • Audio Codes SBC's and SBA's & SONUS SBC's and SBA's • Comprehensive understanding of networking concepts pertaining to Microsoft Lync server 2013/Skype for Business • Managing Lync L1 and L2 operation support team • Planning and Implementation of Microsoft Lync 2013 enterprise voice Integration with Mitel MXONE IPBX • Installation and Maintenance for Mitel MX-One Server (IP based EPABX) and MX-One TSW (TDM Based EPABX) which are telephony switches used for ECN voice services • Exposure in modern concept...

Hello, We have a working FreePBX distro installation working since 2 years. Now we want to do following enhancement in our FreePBX server. 1. We want to use our FreePBX server for internal office text message communication. I know this can be possible using XMPP module of FreePBX. But don't know, how to configure it. You have to do that. On the other hand we want to transfer file as well at the XMPP messaging window. I don't know whether it is possible or not using XMPP. 2. We want to do video calls between extensions. Need to do video conference as well. You have to configure our FreePBX to do so. 3. We are using Bria Professional as SIP client. You have to configure Bria with FreePBX so text messages and video calls could be done as well. Regards, H. Sattar

Hi Harmohit S., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I want to setup click2call on our asterisk based Vitalpbx.

Hi Team, I need someone to setup click2call on our asterisk baes vitalpbx.

We need a "branded" softphone with a modern design . This softphone could be based an already existing open-source project in order to let you work exclusively on "branding" and "design" of the product. The main features we need are: • sip communication, • SIP SIMPLE protocol(for instant messaging) • Branding and "modern design" • Provisioning through API • Presence status (not mandatory) • Video call (not mandatory) We already tested this two open source solutions which work with our pbx, have all features we listed before and both give open source code Linphone: Microsip:

We have two fresh installations of vicidial and goautodial. We would like to configure one to test some outbound voice blast campaigns. We would like recommendations on which product to use for our needs. So Goal is: 1- initial trunk setup to asterisk or sip trunk setup. 2- Setup 2-3 campaigns based in one file txt each with all non agent "virtual/voice blast" campaigns. We have 5 different voice messages per Campaign messages which will be based on a specific value on the list. We have text files with caller information which we can convert to csv format. We would like to have the process of uploading daily lists automatically (via cron or other interface) from a directory where daily files are transferred/deposited during the night. Also have the campaigns start automat...

Want to build web application name ( Call Dashboard ) and integrated with IP-PBX system to collect all logs then generate live report on the page interface. as a below : IP-PBX ----> Live Data collector --------> Database -----> Generate different type of Reports + Display live on Web page interface

HI would like to add an external IP when calls sent from voip switch to asterisk and add and an external ip which would be media ip and sent back the calls to Voip Minutes.

Online Help with implementation in explaining the architecture

We did SIP peer trunking between two pbx, Mypbx and Avaya. from Avaya we are able to receive calls but from mypbx we hear " all circuits are busy try later" message. kindly anyone helps me with this to solve.. regards

want to develop multi tanent Voip server with PBX capability.

Sangoma FreePBX setup on PBX computer

Integrate vtiger CRM with PBX systems, Twilio, and osTicket.

Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

Our asterisk Gsm Gateway software use PCI board gsm modules its Hardware , the software is working fine ...we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

I have a grandstream model UCM 65 10 that reboots and connects the voip trunks very well. however after some minutes, the registration refresh is not working properly. need help: this is the error message in the asterisk logs WARNING[1923] chan_sip.c: Timeout on 896482223-292013357-1450912169 on non-critical invite transaction.

Hi I need an experienced developer with wallboard design , that understands a call center operation and wallboard requirements also reports , logon and logoff features etc. Thasnk

We have Installed the latest version of issabel with asterisk 16. We Also installed Vtiger CRM 7.1.0 and the connector works succesfully. At the moment click'n'call works perfectly and it's possible to pair an extension with a single vtiger profile. We are also able to see a pop up each time we call a number from vtiger interface, This pop-up only says that the extension received the call. We Would like to have the "incoming call" pop-up in order to see immediately who is the caller or in order to open a new ticket directly from the pop up.

Buscamos un programador. Para desarrollar, modificar y agregar nuevas funcionalidades de nuestra central PBX Issabel. Actualmente ya tenemos el módulo call center funcionando. Relación a largo plazo, para tener un programador / soporte mensual.

I have odoo v11 community edition. I am looking for a freelancer who can help me to implement VOIP calls form Odoo. I would like to implement function of click to call in contacts.... I dont have any asterix, freePBX or cloud calls system yet, I do have SIP Trunk lines. I have seen many projects that had integrated odoo with FreePBX system, and I would like to do the same. So in short you will be connecting odoo with our freepbx system. the odoo crm, contact, leads, customers and more, should have click to call function, popup for outgoing call. If you know odoo 11 and you already develop/integrate phone system, please let me know. I need this job to be done ASAP. Thanks

I need expert guys who have good knowlege of opensip and Asterisk

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

Hello, I'm running VitalPBX. Using PJSIP extensions with SIP trunks. All with TLS signalling and not encrypted RTP. When placing multiple calls at once, voice quality is horrible and choppy. I need someone who can look into Asterisk settings for me and possibly do a network capture and understand what is going on.

i need a sysadmin to Install A2Billing + Asterisk + FreePBX

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

I want to create a Voip trading platform (like for example: - See Video ) for VoIP sellers and buyers. The system has to work with Kamailio and Asterisk.

Need someone to configure click2dial to make calls. Calls using softphone and freepbx - done. Odoo is connected to freepbx - done. If you know odoo 11 and you already develop/integrate asterisk phone system, please let me know. I need this job to be done ASAP. Thanks

We’re looking for someone who has extensive knowledge of VOIP services in Odoo with OnSIP. Also good knowledge of VOIP Asterisk server and VOIP integration for Odoo. If you're an expert in this job, please apply and others are not acceptable. Please submit your proposal with your best samples. Thanks

I am looking at implementing Kamailio within my VoIP network and looking for some guidance on ...my VoIP network and looking for some guidance on how to make it work the best in my environment. I have multiple customers that are registering their phones in my private cloud network. Each customer has a dedicated FreePBX within my network. I want the customers to be able to register phones to their specific PBX using the domain name and registration info provided without having to use a Public IP for each FreePBX. In other words, Multiple customers can be routed to the correct PBX with the use of one Public IP address. Like Session Border way I don't have to purchase multiple Public IP addresses and can have Kamailio manager the call routes and voip calls. Please offer ...

We have a newly installed FreePBX server hosted by Cyberlynk. The PBX was setup by our SIP trunk provider. Currently the UniFi phones receive calls but with no audio. The UniFi phones send audio to the other caller fine. We need someone who has experience with Hosted FreePBX AND UniFi VoIP phones to assist with resolving this audio issue. There's not much documentation on the UniFi phones and they are difficult to work with so please only apply if you have this exact experience and can resolve our issue.

Need a individual that can develop me a connection between my asterisk voip server to a crm system i build in house.....

We are looking for a programmer with experience in working with audio/video (A/V) SIP VoIP communications using WebRTC and WebRTC mobile, call managed through a PBX (preferably Asterisks). The first and main engineering task is to establish A/V SIP VoIP communications from a Pine64 SBC running Linux Armbian through a Web Browser to a smartphone (iOS/Android). Asterisks and the WebRTC services are to exist in the Cloud. After the successful completion of the above task, we would also like the to establish a connection from the Pine64 to a windows 10 PC (running the Chrome web browser). The objective of this project is to create a reliable communications path accessible to other developers through API calls at each point. These other developers would develop Web/Node.js, iOS, A...

I must have made the follo... Wednesday, Thursday and Friday at 4pm (4pm) Must activate the following commands asterisk -rx "queue add member Local/28282836@phones to 32212 penalty 7" asterisk -rx "queue add member Local/28282836@phones to 32211 penalty 7" asterisk -rx "queue add member Local/28282836@phones to 32213 penalty 7" asterisk -rx "queue add member Local/28282836@phones to 32214 penalty 7" 3. Script Must be every Monday, Tuesday, Wednesday, Thursday and Friday at 07 (7 am) Must activate the following commands asterisk -rx "queue remove member Local/28282836@phones from 32212" asterisk -rx "queue remove member Local/28282836@phones from 32211" asterisk -rx "queu...

install openvpn and fix the connection issue in asterisk and fix some issue

my provider's server crashed (raid issues) and i need a find gentleman who can install freebpx / asterisk and secure it on a cloud server having ssh access...

...Eremin P., I'm interested to talk to you about our project. We are a small group of professionals that have a few FreePBX solutions implemented in some of our customers and we are developing a web interface that has the ability to show queue statistics in a more dynamic and intuitive way and also have some agents control with the tool. Eventually we would like to implement dialers and outbound stuff but in a more basic way, not yet to implement predictive or so. Also we would like to have a Webrtc phone integrated in the agent panel that they can interact with. So far we are developing the tool using Laravel, Vue.js and MySQL, the application interconnects with FreePBX to get data and eventually the idea is also use the capabilities of Asterisk/FreePBX through AMI and...

Hi We have installed vital pbx 3.0 and we are looking at moving all our clients over from kiero operator to vital pbx. I'm looking for someone to move the clients over one by one. I will advise you the call flow setup along with a zip file for each client for IRV's, welcome messages, hold music, etc You are then to setup each client as a Tennant in vital pbx Along with configuring other vital pbx settings for clients to get the most out of the platform. We as a business will be the first customer to go live on the new pbx. Once we test the platform for a week then other clients trunks can be added and they can go live We have around 100 extention across all clients With your experiance we will also look at other wholesale sip provid...

I have an asterisk server, the calls are droping as soon as the call reaches the IVR, I require help to fix this problem urgently please.

We have a Asterisk/FreePBX-system with 2000 preconfigured extensions. We need a software to run on another Linux machine that continuously monitors the status of the Asterisk/FreePBX system state and triggers email-alerts if the server is unresponsive. The software also need to log in a local SQL DB when a PJSIP-extension/endpoint on the Aserisk/FreePBX changes state. Details of the Endpoint should be logged with the state change and email alerts should be able to be configured on specific extensions and state changes. (I.e Unavailable->Available/"Not in use" and Available/"Not in use"->Unavailable). The software needs to send heartbeat email alerts that it's running and have access to the Asterisk/FreePBX server. All email alerts need t...

We have a single Virtual Phone Number that we want to direct to our Linux cloud server, most likely running Asterisk. We only need a single account so that when the Virtual Number is called then will be forwarded likely via SIP to the servers IP. This will then play a single Audio file of our choosing. Perhaps later we might expand it into a full auto attendant, but we only really need a single message then the user will be forced hangup.

our Gsm Gateway software use PCI gsm modules board as its Hardware we are now changing the hardware from a - PCI gsm modules board- base to - USB GSM module or modem , for that we need to to Modify chan-rgsm module software to adapt the new hardware base

Im getting hacked , to the point its slowing down system I need someone to stop this

**To do without any commercial module** I have a requirement of Asterisk/ FreePBx to make it redundant. for that what my client is looking for is One main PBX at Location A and another PBX box at Location B. In case Location A goes down for any reason then Location B should kick in and continue the phone service for the company. Is this is something you can do? Please let me know.

Required expert in Cisco phones to setup custom features for the phones. We are using Elastix pbx server. Would like to configure phones to auto pickup on speaker phone without ringing.