Asterisk send email phone call jobs

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    10,936 asterisk send email phone call jobs found, pricing in SGD
    VoIP termination solution 6 days left
    VERIFIED

    We ar...caller to enter dtmf. no dtmf drop call) Must have 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 .

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    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.

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    13 bids
    Asterisk AMD 5 days left
    VERIFIED

    Need AMD configured in dialplan for asterisk

    $18 / hr (Avg Bid)
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    8 bids

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

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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 4 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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    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 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

    $37 / hr (Avg Bid)
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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.

    $3866 (Avg Bid)
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    8 bids
    AMD detection in Asterisk 1 day 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.

    $73 / hr (Avg Bid)
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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 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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    Add ADI input to Asterisk frontend. Thank you!

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    3 bids

    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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    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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    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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    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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    helping in Project to finish product.

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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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    you have to install asterisk 1.8 on Puppy Linux make sure asterisk work with sip & iax both also wireguard & TeamViewer you can download OS from here [login to view URL] also if you want you can use Ubuntu or Centos you need to make iso and iso size must be under 500MB i can pay up to 100USD do not ask for more

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    Hi Sourish I'm looking for a Asterisk Engineer to assist our team with day to day management of our Asterisk cluster and programming. Can we chat or Skype to discuss more detailed requirements and your availability. We are looking for 20-30 hours / week. Kind Regards Joel

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    Hi Santosh I'm looking for a Asterisk Engineer to assist our team with day to day management of our Asterisk cluster and programming. Can we chat or Skype to discuss more detailed requirements and your availability. We are looking for 20-30 hours / week. Kind Regards Joel

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    ...necessary, with experience and knowledge within • On premises deliveries of system and installation script • Telephony and SIP (FreeSwitch, Swyx (a German PABX) and eventually Asterisk) • Record keeping, eventually through a PHP framework • Access rights for different users, we work with an access token today Extent We had in the past few years one person

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    Voip expert with PBX (asterisk/freeswitch) and dialers (vicidial) experience to help deploying a special use case related to a voice bot product (not for human agents) A relationship with our low level developers. A long term job relationship is promised to whoever will fit the job

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    I need voip programmer to assist me with some projects in asterisk or freeswitch.

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    Hi, I have a need for an app which can connect to my Freebpx/Asterisk Server I will give further information to any replies.

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    ...semi colon clears text and starts a new section. . two commas mean a delay of 2 seconds before the next piece of text is played.3 is 3 seconds, 4 is four seconds etc.. An asterisk changes the following text to be in a different colour. See example [login to view URL] This text would be typed as follows in the box Costa

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