Asterisk pbx fedora işler
Google Dialogflow kaynağını kullanarak Asterisk veya Freeswitch ile çalışaçak Etkileşimli Sesli yanıt IVR oluşturma.
PBX telefon centrali uzmanı Genel Nitelikler - İleri Düzey PBX Sistemleri ve Voip Gatewayler hakkında bilgi sahibi. - İleri Düzey FusionPBX, Freeswitch, 3CX telefon centrallerinde uzman. - Sip telefonları provisioning yapabilen. - Tercihen linux scriptleri yazabilen. Atomasyon yapmaya meraklı. - Network bilgisine sahip (TCP/IP, LAN, WAN, VPN) - VoIP proje ve uygulamaları konusunda en az 3 yıl deneyimi olan - WAN uygulamaları ve cihazları, internet, firewall konularında deneyimli. - İngilizce bilen. Teknik döküman takip edebilecek düzeyde ingilizce bilgisine sahip - Evinden çalışabilecek. (Home ofis) - Çözüm odaklı olan, analitik düşünme yeteneği gelişmiş, sorumluluk bilinci yüksek, - Yoğun iş temposuna ayak u...
Asterisk santrali c# dili kullanarak ve kaynak kodlarıyla beraber bir yönetim programı oluşturmak istiyoruz. Daha önce asterisk santrali üzerinde çalışmalar yapmış, konuya hakim kişiler ö santrali c# dili ile yapılacak görseller projemize eklenmiştir. Buradan inceleyebilirsiniz. Bu tarz bir yönetim paneli talep ediyoruz. Bu projede bizlere yardımcı olacak ve proje ile ilgilenen kişileri teklifleriyle birlikte bekliyoruz!
Merhaba, 40.000 Kullanıcı kapasiteli, eş zamanlı 10.000 çağrıyı kaldırabilecek Asterisk kurulumu için destek verebilecek bu konuda IP Telephony, Call Center, Voip, Network konularında tecrübeli çalışma arkadaşları arıyoruz.
Elastiks veya İssabel pbx santral üzerinde bir kaç ayar yapılması gerekiyor. Asteriks bilgisi olan uzman arkadaşa ihtiyacım olacak.
Cihaz, kurulum ve bakım maliyetleri olmaksızın; gelişmiş bir santralin (PBX) sağladığı tüm özellikleri; bulut teknolojisi, “Bulut Santral” veya “Sanal Santral” olarak da bilinen ve VoIP ses hizmeti sağlanan İşteSantral ürünümüz ile sizlere sunuyoruz. Üstün özellikli sanal santral ses hizmeti çözümüne; uygun maliyetle ve aylık ödemelerle sahip olmak istiyorsanız, yeni lokasyonlarınız ile büyümeyi planlıyorsanız, şirket içi görüşmelerinizin ücretlendirilmeyeceği şekilde, bulut santral çözümümüzden faydalanabilirsiniz. Uygun Maliyet İlk yatırım maliyeti, santral bakım ve onarım masrafları olmadan kullanıcı başına aylık ücretlen...
Hello there I want to create a callcenter api with Fusionpbx I want to do with this api. 1- Create Callcenter. 2- Add and delete agents in Callcenter Queue. 3- Agent creation. 4- Agent delete . 5- Changing the status of the callcenter agent (Logged Out - On Break-Available) 6- Callcenter queue status.
Yusuf Bey Merhaba, STH lisanslı bir telekom firmasında teknik departman müdürüyüm , şöyle bir sorunumuz var daha önceden çalışmakta olduğumuz aculab prosody s üzerine kurulu voip sisteminden asterisk platformuna taşınmaya çalışıyoruz test amaçlı kurduğumuz server üzerinde denemeler yapıyoruz, şu anda çağrılarımızda bir sorun yok istediğimiz şekilde çalışıyor olmasına rağmen dtmf detection sorunumuz var, ne yaptıysak asteriskin dtmfleri yakalamasını bir türlü başaramadık, bu konuda bize yardımcı olma şansınız var mıdır?Teşekkürler.
Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.
Merhabalar, Lütfen Hızlı Türkiye 'den sadece benimle iletişim kurun. Başarılar dilerim.
Asterisk üzerinde çalışan crm yazılımını geliştirmek için proje desteği istiyoruz yapmış olduğumuz crm yazılımına data arama ve data kayıtlarının tutulduğu arayüz entegrasyonu yapılıcak.
Merhaba Benim bir voip softswitch im var. itelbilling kullanıyorum. Bazı hayal edip istediğim özellikler burada mevcut değil. mesela IP PBX , CALL CENTER , SESLİ ANKET , HOTEL , VOD AND DEAMOND , IP TV. Bütün bu sistemin çalışmasını tek ekranda yapabilecek var mı ? portaone gibi bir sistem istiyorum. detaylarını daha fazla anlatacağım
I have issues with the production server so I am looking someone to fix it asap. I have 2 different applications assigned to the the same carrier and carrier transfers calls simultaneously to both the ip, but right now as the call reaches fusion its disconnecting the call not allowing it to get transferred to asterisk server.
Need help in inbound and outbound etc .
I have a linux server running asterisk running Chan_sip which only support UDP, and can not be upgraded. The asterisk Server communicates to Phones and SIP TRUNK Providers on UDP port 5060 using the LAN Public IP X.X.X.X interface. I need to have some new phones connect to the Server over TCP Port 5062 I need to have some new phones connect to the Server over TLS Port 5089 We will need to have kamalio or Opensips centos 7 to run on the same server as asterisk, and should allow the following: 1) The Proxy should communicate with the asterisk server on UDP SIP port 5060. 2) Listen for incoming SIP TCP traffic on 5062 on the LAN IP X.X.X.X. and proxy this SIP TCP Traffic to and from the Asterisk server on UDP Port 5060 using the interface. 3) ALL T...
This is for a Asterisk and Zoho CRM integration. We need to receive and make calls on Zoho. Also be able to record, transfer, pop up a window with caller contact on Zoho.
API SDK module required with Asterisk using PHP and MySQLwith subscription and credit billing model for users to enable them to add their own SIP trunk and route their own sip accounts and use API through any of the SIP trunk they have to process their IVR instructions on given webhooks. Basic MySQL structure to start with. a) users table [api_key (unique), loginid (unique), pwd, trunk_id , user_type (admin/user) ] b) trunks table [trunk_id, username, password, ip/address , port, protocol, transport_type, did_numbers, status] and other fields and table that is required. Experience Required : Asterisk AGI PHP MySQL Reference API SDK documentation Calls will be made through the voip server using the rest API tht needs to be developed in PHP below is the sample how we will
Hi Pablo B., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I'm installing Kommo CRM in my company and we currently use voip offered by local telephone company. I'd like to configure our voip numbers on asterisk and then Integrate with kommo CRM. Can you do it?
I have 2 asterisk servers A and C A is running on Centos 7 and has a static IP C is running on Freepbx (raspberry Pi 4) and has no access from the outside. We also have B which is running OpenVPN and has a static IP current configuration is A -> B > C -> outside Trunk We need the following to be done 1. A SIP remote extension to log in to A 2. A passes the registeration via B (OpenVPN) and register as an IAX2 extension in C 3. SIP remote extension made the call, A will translate the call from SIP to IAX2 Another thing, you would need to access using Anydesk via my PC to access both A and C due to VPN. Thanks
I need someone who experience in Linux and script development knowledge for Asterisk /freepbx patching. I will integrate existing asterisk PBX to Kommo CRM. The Asterisk/freepbx V18 PBX and kommo are ready, but don't know how to run the script development as below link:
I am looking for an experienced developer to help me design and implement a full Cloud Pbx system with mobile application (example: 3CX). I have decided to use a multiple-server architecture, and I do require specific hardware and software solutions. I also have a deadline in mind, so this project needs to be completed quickly and efficiently. If you believe you are a fit for the job, and can help complete my project in a timely manner, please contact me. Thank you!
I am looking for a freelancer who can help me build a cloud PBX telephony software that is supported by Twilio or Asterisk, using the SIP protocol. This telephony software will not need to be integrated with an existing management platform.
...IP-PBX using Raspberry Pi 4. System works fine internally within the SIM router. But when remote extension trying to log in to my IP-PBX, that's where problem starts. After digging, only to realized that WAN IP for my SIM data provider has multiple LAN IP. My mobile SIM router is given a LAN IP and a shared GSM WAN IP instead of dedicated one, therefor inbound traffic is not available. I need something like. OpenVPN sits in the middle between my remote extension and my IP-PBX. OpenVPN comes with a dedicated WAN IP where my remote extension and my IP-PBX can report to. Inside OpenVPN, OpenVPN would be able to connect them together using LAN IP. Scenario would be, 1. IP-PBX log on to OpenVPN using OpenVPN dedicated WAN IP. OpenVPN will a...
...will provide the pop3 email access to the phone system in order to fetch the new emails and allow the customer to control their emails by reply, forward, delete or mark as spam by selecting options on the phone keypad or by voice commands. Each customer will have an email account on the email server and will be able to access the email address by placing a call to the phone system (based on Asterisk pbx). Each customer will use an authenticate method to authenticate and allow the access to email account through the phone system. Each customer will be able to import up to 5 external email accounts (Gmail, Yahoo, AOL, ... etc) by using the POP3 access and the system will allow the customer to manage these emails with their email account at our private domain. Also, it wi...
Need a short video with some facts about our pbx system, I added a link for an example. it should have good resolution and easy to share on social media sources. Text I will supply.
We are having major issues with our vicidial. Channels not connecting and calls being burned and reps sitting waiting. We are losing money and employees daily and need someone to help us get dialer running smooth asap.
We are looking for someone who can integrate Asterisk in our Laravel system. For each incoming call, a pop up should open with the details matching in our existing database. For outgoing calls we should click the phone icon and call should be redirect to the customer number. You should have prior experience in Asterisk /FreePBX integration with Zadarma or Plivo.
Dear Freelancers, We have a video bridge jitsi installed. We need to achieve the following: Need to be able to login from a raspberry pi with a calling of url. The url call will happen if someone dial an extension on an asterisk pbx. Once the extension dialled from a SIP phone asterisk call a url which makes the raspberry online in the jitsi meeting room. From this time the raspberry is opened the meeting roon and participants can log in to the room. The raspberry pi need to work with a Konftel conference system . So the Konftel will be connected with USB to the rasperry and then the raspberry need to use its camera and microphone in the meeting.
i have laravel app that is managing user and apis . now i want someone who help me to in aws configuration for Laravel and SIP server asterisk ( user mapping with device)
On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. this module had to be compatible with latest freepbx 16 version. - Freepbx Asterisk distribution : - superfecta module to work with : fixed price : 100 $ - ASAP within 1 day.
I'd like the following to happen. A user registered to asterisk will dial the extension 1001. When they dial 1001, they will enter a conference room Another user registered to asterisk will then be dialled and forcibly put into that conference I like this to be done in Node.js, using Asterisk ARI to orchestrate as much of the above as possible. I'd also like the freelancer to share with me instructions on how to deploy asterisk, their config on asterisk and the node.js code too, so that I can recreate their experience locally. Be great for this to be done within the next 24 hours!
Required an outbound calling agent (B to B Sale) who can speak fluent English to sell Telecommunications Services (Cloud PBX, Cloud Call center, Broadband/Internet etc.) to Business entities including small offices. The agent will have to make phone calls from a database to offices and explain Telecom services like Cloud PBX, Cloud Call center, Fiber internet packages to entities based out of Bahrain. The daily leads generated with contact details must be recorded in the company CRM system. The job is target driven with predefined levels of sales leads generation target along with an incentive plan for exceeding target. The agent should be able to speak fluent English, excellent conversation skills engaging with business, prior experience running outbound sales campaign f...
We need to develop web based sip client application that should work on all mobile browsers without downloading any extension etc for browsers . and should work with free pbx etc
The project is a design and implementation of a doorbell using raspberry pi4 with python using asterisk (or another open source), zoneMinder (using ip camera), with 2 android mobile phone, at deferent flour. I need only the CODE of the asterisk (or another open source like: freepbx or fusionpbx), in orderto call and listen rasberry pi4 and android mobile phone, (flour number 1 or flour number 2). I also need a intercome between the 2 flour from each mobile android phone, at deferend flour.
Se requiere free lancer para desarrollo de funcionalidades y aplicaciones en Asterisk
We use Google workspace at work, so all' our contacts are stored on Google contacts. We use freepbx as our pbx and I would like to be able ti have both connected to see the name of Who Is calling via freepbx querying g contacts.
We are looking for an experienced VoIP developer who can design Windows and MAC desktop VoIP applications using our Hosted PBX API. The application will have to be tightly integrated with our asterisk-based PBX and our custom API. Supported functionality will include: Voice calling via SRTP Searchable Call history with access to call recordings and call notes SMS and MMS messaging Read-only access to favorites and BLF keys Read/Write access to personal contacts Visual Voicemail Do not Disturb Call Forwarding We prefer a web application running installable with an Electron wrapper on the client's workstations but are willing to entertain other options.
Hi Need to design a sip based extension portal and forwarding portal. Looking forward to hear from you
Hello, I need some help with a network installation that we made. We have an UDM pro plus a few USW24. The network is working ok, but I need some help with the PBX. The yeastar pbx is not registering the phone. I think I might have something to do with the network because everything was ok before we install the UDM and the USWs.
Build an agent's dashboard as an OMNI channel contact center to handle sales, support, and marketing queries. If someone has a ready solution let me know if you guys can build from scratch and have extensive expertise and skill to architect and build from scratch share with me the tentative timeline and cost. It should support cross integrations with CRM and other systems. I prefer to have it in Angular/.net /Java with MS SQL or Laravel Bootstrap or angular with mysql. Get back to us this is urgent. Regards, David.
On our IncrediblePBX vps : add contacts db source and make connection with growl to get customized notification via Growl send-to source based on SIP line called, and caller details if found in local phonebook mysql db. - Freepbx Asterisk distribution : - superfecta module to work with : price : 50€ fixed - ASAP within 1 day.
Callback widget - Asterisk Hello! I want a price for coding Asterisk to work with a callback function. Simply explained, we will publish a form on a website where customers can request to be called within a minute. When the customer has entered his phone number, it must call one or more pre-entered numbers in the Asterisk telephone. When one of the people being called answers the phone, a voice recording should be played: "A customer wants to talk to you, click 1 to confirm that the call should be connected" When the called person has clicked on the button to confirm that they want to talk to the customer who wants to be called, Asterisk must call the customer who wanted to be called (on the phone number that the customer specified in the form). ...
I need a voice bot for accounting office in greek language. 1. The client say who agent want to connect on 20-30 key words the call go to specific agent or group
Develop a character named Yowie Howie - a friendly, wise cryptid that is a cousin to Bigfoot. - Howie is smiling, has a gold tooth - thumbs up - light grey fur - green eyes - healthy build but not too muscular - has a walking stick - wears a fedora - Maybe winking - full body view Production time frame is two weeks from accepting the job
...structuring: ___ DB structuring & Maintenance: ___ API (creation): ______ API (integration): _____ Hostgator Server Set Up: ___ Hostgator Server Security: ____ Hostgator Server Maintenance: ____ Linode Server Set Up: _____ Linode Server Security: _____ Linode Server Maintenance: _____ Managing Linode access with SSH keys NOT with passwords: _____ Laravel:_____ Vue JS:____ React:____ Github: _____ Asterisk: ______ Trouble Shooting: ____ PCI Compliances______ Please add anything else you would like us to know about your skills and be honest so that we know what projects are best for you as we have many + What is the best per hour rate you can give: ...
necesito me terminen de configurar un servidor interno de voz montado con issabel pbx. por wifi funciona perfectamente las extensiones moviles, pero al quitar el wifi de los moviles se corta y no se escucha el audio creo que solo tengo un problema de puertos. necesito ayuda de uno que sepa.
Allow multiple customer to login my portal to broadcast voice from their own upload list. Another feature require is to check contact list is valid or not. I have my own pbx.
Necesitamos implementar una IP-PBX Issabel desde cero: instalación y configuración remota para 252 internos, troncal SIP y servidor hot standby.
The project is to create a service running on Debian operating system. The service should connect Asterisk PBX, open a socket on port 7891, and send the output of the following Asterisk events: - Inbound Calls: Call waiting in the queue, Call Ringing, Call Answered, Call Hangup, - Outbound Calls: Call Waiting, Call Answered, Call Hangup. The output of each event should be in one line only (Not sending few outputs for the same event). The project includes also a client that should run on a pc, connect the remote server on port 7891 and receive the server outputs.