Asterisk dialplanJobs
Vb6 ivr asterisk dialogic programme
Sonetel & Asterisk Configuration
a dialer for my asterisk box kasdjlasjdlasjdlasjdlasjdalsdjalsdjalskjdalskjdalskjdalksjdalskjdalskjdlksajdlasjdalsjdalsjdalsjdalsjalskjd 100chars
a dialer for my asterisk box kasdjlasjdlasjdlasjdlasjdalsdjalsdjalskjdalskjdalskjdalksjdalskjdalskjdlksajdlasjdalsjdalsjdalsjdalsjalskjd 100chars
Добрий день, Мені потрібно виконати інтеграцію Asterisk+Zendesk - бажано, щоб при надходженні дзвінку створювався тікет, ідентифікувався дозвонювач, і т.д. Чи є в Вас такий досвід роботи? Ми знаходимося у Києві. Бюджет та подробиці оговоримо.
incoming calls on a DID. Caller gets chacked from a MsAccess database for service access and a dialtone to enter a destination number. Call must get transfered to destination and a log of the call length kept in a MsAccess database.
Se dispone de un servidor WebRTC Janus o Asterisk en linea. Se desea obtener una página web con un marco que realice la conexión webRTC a un destino mediante el servidor Janus o Asterisk y establezca la videoconferencia. La página web debe tener tambien un botón para llamar y otro para cortar. La página debe funcionar con sistema operativo Android (...
we are facing issue with vicidial , the incoming call not come , its get hangup after dialing number look like that dialplan is get crupted can you able to help me for this. urgently. earlier this is working fine but there is some issue and one of our admin agent deleted the Var/ log folder to free space and there after we face this issue that dial
...ultraschnelle CRM-System, dass wir zum Herzstück unserer Firma ausbauen wollen. Derzeit ist die Situation folgende: Wir haben auf der einen Seite eine Telefonanlage in der Cloud (Asterisk). Das klappt zuverlässig und bleibt. Auf der anderen Seite haben wir ein Cloud-CRM System. Dieses System muss auf Grund diverser technischer Schwierigkeiten durch unser eigenes
We are termination provider we have a specific solution ...on/off feature Server C: register gsm gateways using pptp vpn and load balance outgoing traffic to gateways based on number of calls For server A we have a2billing and asterisk in mind And for server C we want opensips with gui installed. When we finish we must test call from one of our .
We have an android app that makes video calls from our system to the users mobile phone. It currently supports up to android 7.0 We need someone who is proficient in Asterisk and Linphone Client We would require the app with upgraded compatibility with android 8.0 and 9.0 phones.
I want to make real use of Asterisk Background detection , want a very small simple dialplan and configurations to Use BackgroundDetect of Asterisk
...an open source bounty. There are some who have gotten this to work with other providers: [log ind for at se URL] There are a variety of ways to interact with Twilio's service. The goal is to be able to tie FreePBX to it in a way that doesn't break upgrades to core, etc...Functionality
I have a server running with mysql and php, I am using it for asterisk based server for voice broadcast. I need help to resolve some issue and enhance its features. The out look will be like attached picture. Details will be discussed during chat.
Hi Aqs Y., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I am looking for some auto dialer based on asterisk
We have setup bitrix24 onpremise and we have asterisk-vici installed. We want to use SIP connector. Any one with experience in setting up onpremise Bitrix24 and telephony with SIP connector can respond.
Hi All I have a SIP client on my network, its a Intercom. It registers to my FreePBX/Asterisk solution and makes calls to my mobile phone fine. I want to intergrate Skype because Amazon Echo's can answer/make calls using Skype. The intention is to receive the Intercom call on my home network to an Echo Dot, which I believe will work when the FreePBX
Hi Kingasterisk T., We have a problem with our asterisk configuration. as a brief when call directed to extension queue after ivr. there is silence and no result. caller cant reach to extension. as example we get this kind of logs. [2019-02-14 08:56:15] WARNING[7089] chan_sip.c: Autodestruct on dialog '223205511-3759123182-18707@TcomNextoneSBC4-DRC
I'm new to FusionPBX/FreeSwitch. Can't set up the same thing I've set up in minutes in Asterisk. Need remote assistance/general consulting or just a setup example in accordance with my infrastructure conditions. My FusionPBX is a VM that has only one LAN NIC. It uses pfSense VM as a gateway. pfSense VM has both LAN and WAN links with a public IP on
I Need help with a vicidial server that is not leaving vvoicemail message and is dropping calls in a press 1 campaign
We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.
We have developed a voice bot that understands vernacular languages of India. We are looking for a developer who can help us integrate a user call with our voice b...asks a question - e.g. what are the current offers? The bot responds: Buy goods worth Rs 5000 and get cash back worth Rs 1000. Please get in touch immediately if you know Asterisk well.
I want someone expert in asterisk who can fix my AMD detection in Asterisk, I'm using Vicibox and running Press 1 Survey Campaign.
im looking for someone to make landing page simple nice looking landing page have form and need image
Freeswitch + ASTPP interface Full installation and configuration Trunk Setup Few Rates Configuration Test calls create user dialplan, prefix routing
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
We have a call centre system using Asterisk. With Asterisk only supporting the following call distribution. We need to tweak the queue behaviour to reset agent idle time in two event, after agent has hung up an incoming call and after agent end pause. If you base on the ring stretagy of least recent, we can achieve the agent idle time reset on the incoming
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
hello i need to stream live calls from my asterisk more or so like chansky but i need to stream in demand and to wav files so i can play it in brower i am useing freepbx
must be connect multi Skype user to a SIP-PBX. and call via skype to pstn or mobile number. multi skype user can running on asterisk machine. extensions are calling via each skype-account
We are looking for an Asterisk expert who has experience installing Kamailio or similar software to be used as an outbound proxy for NAT Traversal front ending multiple Asterisk/FreePBX servers. Some PBXs are having load spikes due to frequent device registration/authentication and NAT traversal functionality, so we're looking to install a solution
Add ADI input to Asterisk frontend. Thank you!
Need to develop end to end solution of call center.
i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work.
i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work,
I want someone expert in ASTERISK and freeswitch and ios to send calls over bluetooth but to use whatsapp or viber instead of gsm dialing to be smart enough to tell asterisk use whatsapp when dialing (send calls direct to whatsapp) android or iphone whatever they can fix
Trabalho há mais de 25 anos em telecomunicações, estou procurando desenvolver um sistema de interfonia baseado em asterisk com acesso através de app (soft phone)
Current software versions: FreePBX: [log ind for at se URL] Asterisk: 13.11.2 Project Description : . After a call, our Call Center Agents will transfer a customer call to an extension where the survey should be executed. Survey will consist of 2 fixed questions. We need to be able to define the Survey number and Name, then record the questions and select valid
1) Set up Asterisk. You will get a machine with Linux OS of your choice. 2) Install UniMRCP and UniMRCP Kaldi Plugin. Configure SIP line for inbound calls. We will provide the URI to configure in the plugin. 3) Configure an Inbound call plan that does the following for inbound calls: (a) Prompt the caller to punch in a 4 digit code. (b) Post
Hi, The goal is to establish an asterisk server and freepbx 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
...config file Sip pbx ip address/port 5060 Sip User/Pass Operator Calling number Time to auto answer Relay Unlock period Sip client must have VPN configuration to connect to Asterisk PBX over internet. Hardware We use raspberry pi3, mic- in we can use over USB sound card Aux out use over PI3 audio out connector 3.5mm stereo jack GPIO22 trigger in (start
Hello, We are looking for someone to deploy a FusionPBX (Freeswitch) for us from scratch we even need everything to be migrated to the new PBX from our Asterisk.
