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    12,279 asterisk call forward callerid jobs found, pricing in USD

    I want to make real use of Asterisk Background detection , want a very small simple dialplan and configurations to Use BackgroundDetect of Asterisk

    $166 (Avg Bid)
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    ...an open source bounty. There are some who have gotten this to work with other providers: [login to view 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

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    php mysql schedule creation 5 days left
    VERIFIED

    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.

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    Hello everyone, I’m seeking an open cart plugins developer to build a simple plugin for me as follows: The plugin is made for our shipping company and is intended to forward open cart orders to our website through our API ONLY IF the order was confirmed and has our company selected as the shipping method for the order. The plugin should do these effects

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    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

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    We have set up a landing page on our website which send people to a Mailchimp landing page to capture the details for a competition. We want them to be able to SHare this with a friend and receive additional entries in the competition. The new referred person is to see the origonal promo page: [login to view URL] (you can go here and follow the links.) We have this setup on mailchimp. Work ok un...

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    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.

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    ...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 server

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    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

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    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

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    I Need help with a vicidial server that is not leaving vvoicemail message and is dropping calls in a press 1 campaign

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    ...integrate a user call with our voice bot. Here's how the expected user experience is: When the user calls, our bot greets the user with a welcome message. The user 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.

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    ...integrate a user call with our voice bot. Here's how the expected user experience is: When the user calls, our bot greets the user with a welcome message. The user 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.

    $2858 (Avg Bid)
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    AMD detection in Asterisk 3 days left
    VERIFIED

    I want someone expert in asterisk who can fix my AMD detection in Asterisk, I'm using Vicibox and running Press 1 Survey Campaign.

    $54 / hr (Avg Bid)
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    I would like to forward, Rewind button to hls player control. You can use any plugin for this . MP4, M3U8 support should be possible in HLS player .

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    im looking for someone to make landing page simple nice looking landing page have form and need image

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    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

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    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

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    Asterisk Expert 5 hours left

    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

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    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

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    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

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    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

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    ...And press on/off button. When user try to call someone in standart android call area, our app forward this call to our choosed VoIP application. And automaticly call from there. There is an application similar this "Premium Dialer". But this application only forward to VoIP app, not start call. Ex: Forward to skype and write num...

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    Finance foreign exchange ,futures,forward ,derivative multiple choice simple questions

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    Add ADI input to Asterisk frontend. Thank you!

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    Need to develop end to end solution of call center.

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    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work.

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    i need help fop2 its installed on a standalone (separate) server than Asterisk Freepbx i can get the recordings to work,

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    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

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    Current software versions: FreePBX: [login to view 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

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    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

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    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

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    ...source code in python and auto start with raspberry linux SIP client must call just one number that we insert in code, over GPIO inputs trigger sip client start call. (Gpio inputs must configurable in code) Sip client must have to play audio file that inform user for call. When press button and trigger GPIO input must play AUDIO FILE 1 ( calling

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    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.

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    ...python and auto start with raspberry linux SIP client must call just one number that we insert in the code, over GPIO inputs we trigger sip client to start call. (Gpio inputs must configurable in code) Sip client must have to play audio file that inform user for call. When press call-button and trigger GPIO input must play AUDIO FILE 1 ( calling

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    I would just need to make sure that the script is able to copy the contents of all incoming messages in two telegram channels and forward the messages into a 3rd channel/chat on Telegram. So in Summary: I want a bot by which i can get all the posts instantly to my Own Channel where i am Admin. I want to get all the messages instantly when it is posted

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    ...want to develop a native CallerID spoofing iOS app. The App allows you to use the Caller ID of your office number anywhere in the world. The technology in the background is already running. there is an api for all functions. the calls are made by my phone server. the iPhone only needs to dial a local number. No voip. only call via gsm with an existing

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    Add the ability to enter sip trunk credentials to asterisk

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    ...calls )from Server A to Server B. Server A = Asterisk server Server B = Asterisk Client server Explanation of the scenario: 1. server A ( asterisk server, with static IP) receiving VoIP calls , with sip protocol, using G711,G729 and/or G723.1 codec and sending calls to Server B 2. Server B ( Asterisk server with PRIVATE NETWORK IP), receiving calls

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    Hi, i need an app that will read tipstertube picks and forward them by email

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    Build Re-seller Front-end for Asterisk Phone system to allow total control of asterisk via Html5 for management and configuring. To be about to add sites exts phone numbers etc

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    Build Asterisk Phone system for re seller with html 5 front end for management

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    Hi mula9211, I want to check if you are open to work at regular hours for us (much more hours than I propose now). We exist for over 10 years in NL and we have an asterisk based cloud PBX. We sell this as a service to companies and resellers. It is based using our own heavily customised linux distro. We also want to experiment with jingle (xmpp client)

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    fail2ban config and fix hacked phone.

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    ...telegram channels and forward the messages into a one or more channel/chat on Telegram. 7. Should have an option to forward (Keep or remove) (Picture, Video, Link and Document) 8. Should forward messages from Forward Channel1 to Receiving Channel1... (F Ch2 to R Ch2) ... (F Ch3 to (R Ch3). individually so on. 9. Should not show the FORWARD CHANNEL's na...

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    Hi Viktor S., I noticed your profile and would like to offer you my project. We need to install and configure asterisk/A2Billing for voip wholesale. Please let me know, when you can be available to discuss the details. Thanks, Nestor.

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    ...are in system only that can route with ivr or ip . Now i want make change without disturbing other dialplan just only one prefix i route to call center or IVR so all calls of that prefix did will forward to there .rest all system will work on previous dialplan [public] exten=>_XX.,1,Set(DID=${EXTEN}) exten=>_XX.,n,Set(CDR(userfield)=${DID}) exten=>_XX

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    helping in Project to finish product.

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    Pretty simple...we are having a social event and the theme is fire and ice. I have a basic design but want some cooler that really pops...feel free to use what ever cool imagery or graphics you want...I want something crisp and cool and fashionable....its a high end ritzy event but not snobbish.. all the content needs to stay the same...we may be doing 3 or for invites for the same event like upda...

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    ... data spike at 3pm cst and then silence need assistance please.. could be log files full ? dont thing so login as: root root@pbx:~# asterisk -rvvvvvvvvvv Unable to connect to remote asterisk (does /var/run/asterisk/[login to view URL] exist?) root@pbx:~# df -h Filesystem Size Used Avail Use% Mounted on rootfs 40G 35G

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