I have FreePBX has inbound trunks from providers and outbound trunks to customers. I have one customer that if his trunk is down it would dial his cellphone. This should be a simple custom dial plan but I'm sure there is someone out there that can have it done in less time that I can figure out. This is FreePBX distro so any custom asterisk configs must be put in the right place so GUI will not over write them. There are multiple customers on this system so this must be for this customer only. Thanks for looking at my project.
need to interconnect vtiger with asterisk or with voipswitch
We are looking for a high level VoIP Support Engineer to assist with our growing install base of hosted Asterisk / FreePBX deployments. Regular on-call evening / night / weekend coverage will be required, as well as daytime availability to help with high level issues. Phone support is required and native level English (UK or AU/NZ English is fine) is absolutely essential. We envision a weekly fee of some sort, with a possible per ticket bonus, as being the most appropriate model given the coverage hours needed.
To build a app for communicating securely that people can download then call or message other users with encryption securely who have downloaded the app . online registration user names or numbers payments dashboards admin/user
Feature of advanced callhandling of mobile calls. Prescence, and rules. Swyx, Mitel phonesystem. Expert. Videocall is always on when you work. Price 18 dollar/hour.
PhoneSystem PBX application made in Xamarin. Development of app with new features. Understanding of database communication. MS SQL-database. Experts only. Good track record. We have a team of experts. The price is 18 dollars. No more or less. Time is 4-5 mounth 15-20 hours week.
We have a project which needs a portal to manage and monitor the call from Asterisk Server to be developed. Modules needed are given below 1) Should be able to Add Users-> Sign Up -> Login 2) Users will be assigned DID(Phone Number, TFN) 3) CDR and Current Running calls report per DID(Phone Number, TFN) 4) Call Forwarding and Priority Define 5) CAP and Conversion based on duration of call
I need to make calls to external lines through granstream gateway that use analog line. I only need a little orientation from someone who know that.
Buscamos experto en WebRTC y PHP, con EXPERIENCIA DEMOSTRABLE en WEBRTC. Necesitamos un experto en WebRTC para un proyecto que estamos desarrollando. El proyecto se basa en unos paneles web a los que cada usuario accede mediante usuario/contraseña montado todo en PHP. El servicio WebRTC de llamadas está conectado a un sistema Asterisk.
For an online telephony project i've almost finishedd, we are looking a WebRTC and PHP expert: We need a WebRTC expert for a Project we are developing. It’s a web panel based Project on PHP/WebRTC where each user enters by user/password. The WebRTC dialing call service is connected to an Asterisk system.
Mirtapbx Multitenant PBX will work on our plant; By connecting via AMI we want to manage the events from the interface. We want to post event url via GUI. Example process: Extension ringing Extension Call Answering Extension Call Termination We want to POST values like CallerID Call Unique in these events. We want to manage these operations from the relevant interface. Please apply for those who have Asterisk - AMI knowledge.
As a multitenant asterisk, we want to convert event data to post data for CRM integrations via AMI. We want to do reading the data. Exam: Ringing agent, call start, call end etc.
...click. 2) When the user clicks on this button, a small popup dialog will be opened inside the browser tab and a call is initiated between user's browser through WebRTC. (Normal permission to access to microphone will be asked to the user, this is part of browser) 3) A call is directly established between browser through Plivo WebSDK () and SIP extension in our IP PBX using SIP URI. 4) This dialog will display the current status of the call which displays Call Initiated = "Please wait while call is getting connected...", Call Established = "Call is successfully connected." Call Ended = "Call is successfully completed. Thank you for having a call with us." Call Failed = "Unable to connect the call, please try again later." 5) ...
I would like to hire someone who have knowledge in writing API or code for my custom agent CRM. We have design the Custom agent CRM in .Net looking the below API for elastix PBX 1.) Total active call and waiting call 2.) Number of agent login and call status 3.) DND/Break From agent panel.
Hi there, we are looking to built our class 4/5 soft switch for our VoIP means we will need complete reporting and pbx functions. I can also show what I am using at the moment to help you understand what exactly we want. Please do not bid or contact if you are a part timer or have less then 3 years experience. It is ongoing project with guaranteed work. People who already worked on class 4/5 switches know what exactly we want. Any questions kindly do not hesitate to contact me. Regards.
Install free-pxb on a debian 9 Add gsm channel + skype channel + Phonebook + calendar + Wake Up + conference room You will agree when you start work that will be completed within a few hours Working hours from 09:00 to 18:00 GMT
Hi, We have an installation of FusionPBX. We have some basic requirements to assist us to integrate into our CRM. We do not need an integration developed, we simply need examples of the best practices to make the API calls we need. Firstly a prerequisite is that all calls must be recorded. Extensions are set to record all. 1a. API call to originate an call from an internal extension t...call to originate an call from an internal extension to an external phone number 1b. We must be able to dynamically set the external callerid through the API call 1c. The local device must auto answer. We can confirm the following header works with our devices (sip_h_Call-Info=;answer-after=0) 2. API call to hangup all active lines on an extension 3. API call to view all active calls on the ...
I have been using my own class 4 softswitch for about 3 years now but decided to change to a rented voipswitch. Unfirtunately am struggling to understand it fully. I need help with configuring expecially, getting the pbx module to work. Need to understand fully how voipswitch pbx works and get it working.
Hello geeks, we need Integrate Asterisk HubSpot and Asterisk Laravel custom CRM, I`m ready discuss project derails feel free contact me , Best Regards GS
I need an end user interface to manage a control basic features on Asterisk using FreePBX. The user should be able to create ring groups, control inbound routes, control call forwarding, control DIDs, view CDRs and check voice mails.
Build a Reporting System , based on Vicidial database architeture, (call_log) for the calls made outside Vicidial Crm (so manual calls) , and (carrier_log) to see if actually the call went through They get stored in call_log table with structure as in () attached The UI should be based in AdminLTE The report should be similar to vicidial reports that ...jpg) attached The UI should be based in AdminLTE The report should be similar to vicidial reports that use Perl, to minimize database overload , the report structure should be as in (screenshot_5) , The db_connect and language file should be separate , There is the need to alter the call_log to store the sip result too No bid if you dont have previous experience with vicidial and asterisk Thanks
main iso link https://www.freelancer.com/users/l.php?url=http%3A%2F%2Fdistro.ibiblio.org%2Fpub%2Flinux%2Fdistributions%2Fpuppylinux%2Fpuppy-5.2.8%2Flupu-528.005.iso&sig=f8bb0783ca9a793504be0860f0aa7c237556055768cc030588618ea121ca50d6 you have to install fastd asterisk openvpn teamview 7 Google Chrome thats it if you can please bid i will boot usb if u did not made before or u dont know how to made iso please dont bid price FIXED under $100
Hi bro, I want to configure, remote VOIP over VPN, I have on my local network "MyPBX" router, WRT router (OpenVPN client), I also have a VPS Centos7 (OpenVPN Server), I want people from the internet to make calls using my PSTN lines throw "MyPBX" router, I want to implement two things: 1- lin...VOIP over VPN, I have on my local network "MyPBX" router, WRT router (OpenVPN client), I also have a VPS Centos7 (OpenVPN Server), I want people from the internet to make calls using my PSTN lines throw "MyPBX" router, I want to implement two things: 1- link remote branch VoIP phone to call extensions in HQ and to use local PSTN in HQ, 2- any one can use sip softphone to link to the HQ PBX, and to call extensions in HQ and to use local...
Hi kingAsterisk, I noticed your profile and would like to offer you my project. We can discuss any details over chat. The system make calls and leave telephone message, reply to the question automatically. Analyse total number of phone calls you made. The Asterisk FreePBX or PBX solution, they don’t have the same function. (1) To build up a list phone calls you made by Excel or CSV (2) To upload voice (3) To create a campaign In addition, another function: IVR schedule The specified function is transfer all incoming calls and transfer all calls to the preset telephone number. When the customer ring up and push the keypad which shows his required. Telephone will response to a transfer calls to the telephone number which one is best for cust...
Hi Ivan V., I noticed your profile and would like to offer you my project. I have a zycoo coovox u50 ip pbx. need to install and get up and running with Zycoo UC Pro. We can discuss any details over chat.
Hi Ambiorix R., I noticed your profile and would like to offer you my project. I have a zycoo coovox u50 ip pbx. need to install and get up and running with Zycoo UC Pro We can discuss any details over chat.
IVR Requirements Install odoo-asterisk connector in odoo backport modules as necessary from odoo 10 to odoo 9 configure asterisk for inter-operability with odoo Install and configure webrtc via odoo and asterisk install available odoo webrtc module configure webrtc module to route through asterisk configure asterisk to accept and route webrtc Configure auto-dialing groups in odoo using contact, leads, or any other editable/custom group. Add calling, sms, multimedia and voice broadcasts to odoo follow up options Optimize server configuration for reliable high audio and video quality configure settings for sip, users, codecs, etc for best quality configure PSTN connection(s) for best quality Enable chat via soft-phone Configure asterisk/odoo ...
Требуются очень хорошие знания по Астериск. Задачи: Создать концептуальную модель следующей системы: -получение номера телефона звонящего абонента (звонок идет на zoiper, а номер нужно показывать в веб интерфейсе приложения открытого на ном же компютере), автоматически поднимать трубку на Зоипер при нажатии на кнопку поднятия трубки в приложении (через интеграцию астериск) , логин в астериске/зопере во время логина в систему, настройка очередей обзвона через приложение
Integrar o pipedrive por API com asterisk PBX para fazer Ligações via pipedrive registrar status das chamadas agendar atividade de ligação quando chamada mau sucedida
I need an Asterisk expert with some experience . There are 3 phases : Planning , Install and configure then Maintenance for 6 months. I want this to be an hourly and fixed price job both that can be agreed upon .
we need a skilled team/idividual to help to port this module from Odoo 10 to 9 and extend the features to truly bring the best of asterisk, kamelio, webrtc and multimedia call center solutions
Hello, I have an FXO gw SPA400 behind a firewall /home router. It has a local IP : 192.168.1.x. The asterisk server has a public IP address. I have forwarded UDP ports 5060 and 10000-20000 in the router to SPA400 IP address. The SPA400 can register to asterisk server. The problem is that: 1) huge delay (more than 20 sec) for outgoing PSTN calls. There is no big delay for incoming calls. 2) PSTN called/caller are unable to hear anything apart from the ring Please apply only if you have worked on similar setup before. The project will be considered as successful when both issues are resolved. Thank you.
Hello, Our Asterisk PBX application went down on our vps. When it came back up we got this error message: "action could not connect: can't connect to local mysql server through socket '/var/lib/mysql/(2)" ONLY experienced Asterisk techs should respond, thank you.
i need to create an ajax api call from my TLD domain here also the "info" part needs to act like an asterisk subdomain without a wildcard so when you refresh the "info" page for example https://sim.psim.world/showdown/info=?12345678910 it changes regardless of where you search like https://sim.psim.world/showdown/info=?46589645764625
Seeking a senior level developer with deep experience building custom IVR systems for customer support and open ticket management. . - installation, configuration, and maintenance PBXFREE / Asterisk based solutions - integration VoIP systems (Asterisk) and CRM - developing IVR and application (incl. AMI/AGI based) for Asterisk - Twilio integration experience The main idea is to identify our customers by caller id, check to see if they have open tickets and connect them to proper agents or resolutions using speech recognition, keyed responses and or knowledge base. Attached is a more detailed description of how the IVR should be setup with our current CRM. We are currently using FREEPBX but its very limited. Please describe your experience building custom t...
I need to create a branding proposal for IT Voip Solutions and AC Global Tech Solutions. The type of service are: IT Service, IP Telephony, IP PBX, GPS Services and Online Backup Data. For that, we need to create brochures and flyers in english and spanich.
Build a Reporting System , based on Vicidial database architeture, (call_log) for the calls made outside Vicidial Crm (so manual calls) , and (carrier_log) to see if actually the call went through They get stored in call_log table with structure as in () attached The UI should be based in AdminLTE The report should be similar to vicidial reports that ...jpg) attached The UI should be based in AdminLTE The report should be similar to vicidial reports that use Perl, to minimize database overload , the report structure should be as in (screenshot_5) , The db_connect and language file should be separate , There is the need to alter the call_log to store the sip result too No bid if you dont have previous experience with vicidial and asterisk Thanks
hi, we need asterisk support. only experienced person can apply. thanks.
We have a Chinese made HSUPA dongle, which has voice, sms, internet and storage features.... it is not connecting to local service provider via sim. we need it to the our local gsm network so all features can be used in asterisk... we believe it may be suck in WCDMA mode... we just need the sim to connect to network in GSM mode and be able to run multpl edongles on the same system...
Integração entre o sistema de telefonia Asterisk com uma base de ERP para pesquisa de dados de usuarios
need a click to dial application for windows that uses AMI examples below this has to work in programs and web browsers which is why i'd rather it be a windows program the obviously configuration variables must be editable
Hello, I have an active project for asterisk. Would like to take a look if you interest on it ?
We currently have an application under WebRTC architecture. We need to transform the audio/video calls compatible with Asterisk IPBX with recording included for: Audio and Video calls and recording.
i need a os Ubuntu and puppy u can make live CD you have to install some pakage openvpn asterisk php teamview 9 fastd ntp iptables iso size must be under 450MB if you can make it please bid
I want a VOIP PBX with a capacity of six extensions and call forwarding and also generate DIDs . Ideally knowledge of XML & VOIP or superior skills will be preferable.
We do what we want ? We we want to the best of our knowledge. We don't know about us, please tell you. Direct us. We want a complete Telecom solution. So in asterisk that can do everything we want. For example, centos + asterisk + php + nginx + mysql + a2billing + freepbx multi-tenant and other service components to the server must be established. To run mysql on a separate server for the web site hosting services on the server will work on a backup basis. Asterisk on a separate server and other services will work on a backup basis. we're thinking of using VMware ESX virtualization software. At the same time security is very important for us. We want to know suggestions and what you can do about security. For example, VPN with SMS authentication entered e...
I want to build a VoIP telecom company. We spoke yesterday. I think asterisk is the best for the job. all the properties of the Asterisk server to be managed via the panel I want. like for example the combination of a2billing and FreePBX. How do we work ? The long-term mean is constant if I want to work with a person who knows. Please tell me what you can do. For example, a snapshot of the database as 2-sided so redundant (cluster) to work as I want. for example, I want to place the server into the 5 continents of the world. These servers are geographically closest to the customer and connect it with the SRV record to a single domain for voice services provided over the server I want to be. The whole structure have to be redundant. Very good attentive visual theme you need to...
The goal is to establish an asterisk server for the purpose of voip termination in GSM network. The freelancer has to do the following: - Install asterisk server and configure it to operate with Huawei usb dongles (modems) having prepaid simcards to connect to the GSM network. - Be able to attach more huawei usb modems to the server if needed. USB hubs will be used to attach the usb modems. - The server shall be able to terminate voip calls comming in codecs g729 and g723. - Send ussd and ivr to each sim card to topup and credit check. - Display how much credit is available in each sim card. - be able to listen to the channels if they are busy. - If a channel is busy the call shall try automatically the next channel and so on till it finds ...
