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    11,393 asterisk video call quality tugasan ditemui, harga dalam USD

    Saya membutuhkan yg paham dan pengalaman utk melakukan setting/configurasi asterisk saat ini sdh terpasang akan tetapi masih mengkonsumsi cpu server boros sekali rencanaakan dipakai utk call center dg jumlah agent 100 orang.

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    ...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

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    VoIP termination solution 5 hari left
    DISAHKAN

    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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    Asterisk AMD 3 hari left

    Need AMD configured in dialplan for asterisk

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    8 bida

    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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    ...solution working (either in screen sharing, video, etc...). I realize this isn't how traditional projects are run - this is similar to an open source bounty. There are some who have gotten this to work with other providers: [log masuk untuk melihat URL] There are a variety of ways to interact

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    php mysql schedule creation 2 hari left
    DISAHKAN

    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

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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.

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    AMD detection in Asterisk Tamat left
    DISAHKAN

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

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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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    I need someone to make calls for a real estate investor company You must know now to use mojo or ready to learn Speak good clean English with little to no accent what 5 days a way at 2.30 hourly. I

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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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    Serve as the first point of contact for customers and injured workers, delivering excellent customer service. Answer inbound phone calls, identify and resolve caller needs, and provide information.

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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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    9 bida

    Add ADI input to Asterisk frontend. Thank you!

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    3 bida

    Need to develop end to end solution of call center.

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    freepbx FOP2 Tamat left

    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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    freepbx fop2 Tamat left

    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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    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)

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    Current software versions: FreePBX: [log masuk untuk melihat 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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    Call center Tamat left

    Outbound Account is hiring excellent and fluent English speakers for live transfer calls for qualified Customers - it's commission basis not asking for hourly pay please . -sales experience is preferred - fixed shift from 9AM till 5PM PST( 7 working hours - one hour break ) - we provide DDV ( dialer & data ) Waiting for hearing from you :)

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    Add the ability to enter sip trunk credentials to asterisk

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    ...main goal to minimize the BW in client side with good quality of voice. We need some kind of bandwidth compression system ( up to 60-80% than usual SIP 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

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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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    ...recuerdas de mi hace mucho tiempo hice un trabajo contigo sobre asterisk, fue recibir pagos automatico usando el asterisk :) Si todabia estas trabajando como freelancer, estoy interesado en en tener ujn sistema parecido a [log masuk untuk melihat URL] No se si lo conoces? lo estamos usando para call center, bueno cualquier cosa em dejas saber, espero estes

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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/[log masuk untuk melihat 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 [log masuk untuk melihat 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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