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    16,321 asterisk web sip 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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    asterisk and onfig Spa 8800 6 hari left
    DISAHKAN

    we are looking for the person who van config asterisk and spa 8800

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    ...are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers, the upstream CDR reports are not suitable for processing the clients billing for

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    Asterisk Server rebuilt 6 hari left
    DISAHKAN

    rebuild asterisk server using backup files sip and dialplan database AGI (python) restore all default functionalitys

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    Asterisk to another server 6 hari left
    DISAHKAN

    i have arunning asterisk server i need to move it to another server it has a database uses python for some application process no changes just move from one server to another

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    Need Install asterisk+a2billing in VPS servers. If you dont know what is that, dont take my time, i will block you

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    ...migrated to a Cisco 3945 router and it does not work fully. I have: Cisco 3945 with Voice license 1 x 4G Samsung phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over

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    Hi, we need freelancers who are expertise in asterisk or [login to view URL] and have sound knowledge of IVR and other communication features. Candidate should have vicidial expertise too. We will need for 8 Months work, include your monthly price in bid. thank you.

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    Install in vps asterisk and a2billing

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    I'm looking for a tech who already has completed a Ringless Voicemail drop system. We will target users in US only so will calli...like Twilio, or FreeSwitch etc..If you Already have completed a Ringless voicemail drop system please bid. Please include the past project info. Thanks Skills required: Asterisk PBX, Telecommunications Engineering, VoIP

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    We have twilio and need someone to seup the Elastic SIP Trunking and make the phones dial out and ring... via Ciscso SPA540G and obihai

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    Asterisk Migration 2 hari left
    DISAHKAN

    Asterisk Amazon Ec2 Migration to another zone

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    ...directly on the number you want to transfer a call to directly in our system via the browser. Our agents use either snom phones or Bria softphones. NOTICE that we have an asterisk 11.25.3 server today, where we have setup both agents, queues, extensions etc, which must all be working after this change. There must not be any downtime in connection with

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    Resolve small bugs 2 hari left
    DISAHKAN

    Hi guys, I have an android app which is already completed. It has few bugs on few phones. Can you resolve them? Are you aware of SIP protocol?

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    i need someone experienced about issabel (formerly elastix) to set up an maintain our vps based sip switches. serving to multiple clients using multiple sip operators,trunk or sip user based.

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    ...CAPITALS. 3. If you have any notes of your own, please write them at the end of the line, and highlight in blue. /* Important: Please write any notes in between the slash asterisk parenthesis like this one. Failure to do this will ruin the rest of the code. */ 4. Remember that this is an iPhone app, so space is often limited. Please try to keep

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    Senior Asterisk Specialist 1 hari left
    DISAHKAN

    To integrate new FreePBX / Asterisk into network with Apache and MySQL Servers. For inbound call centre that uses WebRTC to connect agents over their browser to the system. Must have excellent knowledge of Asterisk with WebRTC.

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    I need an Avaya IP Office expert who is familiar with SIP Trunk configuration with fax server products. Currently we have the licensing installed for 4 SIP Trunk Connections, but the Avaya IP Office is responding with a SIP "Temporary Error".

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    Hello, I need someone to install a working version of GoAutoDial v4. The instructions are here: [login to view URL] I will provide the VPS required and SSH access. Requires: PHP, MYSQL, COMPOSER, ASTERISK knowledge.

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    ...should have a universal form of support for mjpeg H264 formats.h264+. H265+ support IPv4 / IPv6, TCP, UDP, RTP, RTSP, RTCP, HTTP, HTTPS, DNS, DDNS, DHCP, FTP, NTP, SMTP, UPnP, SIP, SNMP, PPPoE, VLAN, 802.1 x, QoS, ONVIF Profile S & G, and to integrate via the SDK P2P, video Analytics modules [login to view URL]

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    I need you to develop some software for me. I would like this software to be developed. Vicidial, Asterisk, Stripe, Voice and SMS API

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    pfSense Expert -- 2 13j left
    DISAHKAN

    ...will be defined by BB1, BB2, and BB3 - We have 3 VPN cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the Mikrotik there are the agents What we need? Is to regulate the traffic, whereas for example

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    MikroTik RouterOS Expert 13j left
    DISAHKAN

    ...will be defined by BB1, BB2, and BB3 - We have 3 VPN cloud VPN servers based on IPsec/SSTP protocol, later on will be defined as VPN1, VPN2, and VPN3 - we also have 3 cloud Asterisk "Elastix Call Center" servers , later be defined as CC1, CC2, and CC3 - Behind the Mikrotik there are the agents What we need? Is to regulate the traffic, whereas for example

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    I have some extensions on an Asterisk system that are not working. I need someone to check them and fix them.

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    8 bida

    Hello, We have multiple asterisk boxes that need maintenance, often passwords reset, names changed, etc. This is an on going contract to support our customers.

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    I have third party Linux Server, SIP/VOIP, when an IP phones registered with server and are assigned with extension number like 501,502 etc, now these phones make communication with each other by dialing their number. What I want to do is, remove the IP phone and use Raspberry module, and it should work as phone, and connecting push button switch on

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    ...installed and got a running asterisk 13 from debian distribution. Added asterisk-gui and got that to sort of work but getting frustrated with what looks like sip/nat issue and that I couldn't find a way to be able to call between local extensions. It's mainly for personal use, I wanted a simple PBX that could handle a small group of sip devices that could c...

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    ...server and then have all SIP traffic route over the VPN to allow 5060 traffic out of Dubai which blocks SIP on 5060. Install IPSEC on the Ubuntu server (currently in production hosting web app) Configure VPN Ensure only SIP traffic flows over VPN lock down the entire solution so the SIP traffic can only go to one IP address our SIP service This is a re...

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    Need someone to integrate m...auto opening the lead/customer/opportunity. If you know odoo 11 and you already develop/integrate asterisk phone system, please let me know. I need this job to be done ASAP. If the price is right and you showed great capabilities in odoo and asterisk phone system, I will have a lot more work for you in future. Thanks

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    I am looking for Asterisk Expert to work on my project. More details will be shared on chat

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    19 bida

    I am looking for Asterisk AGI expert to work on my project. More details will be shared on chat.

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    8 bida

    I want to connect calls through our Asterisk that is already implemented and being used. The task is to use Asterisk to connect two external phone numbers from our API through the system and make a conference call / bridge. This will require good understanding of how Asterisk works and being familiar with freePDX. This task will require writing some

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    Make calculator for website SIP calculator, retirement calculator ,loan calculator

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    I have a, Asterisk, FreePBX and A2Billing Server installed in Google Cloud on a Free PBX. I need help to configured it so that I can bill departments in my I have already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers

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    I have a, Asterisk, FreePBX and A2Billing Server installed in Google Cloud on a Free PBX. I need help to configured it so that I can bill departments in my I have already installed A2Billing asterisk server and set up a Sip Trunk to my VoIP provider. My internal free PBX extensions are working, however, I need help in configuring the A2Billing servers

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    I need 4 assemblies to be created in Profile Builder version 3: They are: SIP wall (including Internal and External Panels), Concrete Waffle Slab Floor, Mid-Floor and SIP roof. Each Assembly will be treated as a Milestone (the value will depend on difficulty). Outline of each assembly will be provided, but there is a lot of detailed information (both

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    I need an asterisk setup which will be driven by dialplans in mysql DB will be on a separate box from the asterisk. 2 or more asterisk boxes will share the same db. Web interface is also needed Create account, edit profile, add rate, delete rate calculate rate based on some other tables custom cdr reporting etc More details at [login to view URL]

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    Put asterisk and prevent sign up page from being submitted if fields are missing

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    ...purchase. Project Pitch We would need an experienced UX/UI Designer to create an appealing visual mockup for our upcoming new digital tool: the Social Impact Profile (SIP). The SIP will be a user database and part of our existing website ([login to view URL]). With this tool we want to offer our customers the possibility to create their own user account

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    we wanted to setup a bpo in bangalore befo...in bangalore before start will setup the server first and test. We have one server with us, which needs to be reinstalled the latest version of linux, download and install asterisk on the system and test with configuring the gateway provided. I want a technician to come to my place and work on the server.

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    NaughtyTales Tamat left

    I have written erotic stories for years. Some in which I have published and others in which I have written for other ...erotic stories for years. Some in which I have published and others in which I have written for other people. I love when I get in the groove and just let my mind wander while I sip my wine and free my mind... I'm sure you will too!

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    We are looking for an experienced Asterisk developer who may have already designed a IVR solution for running a psychic line via VOIP. The system needs to be able to do the following: 1. Auto answer calls 2. Allow user to pay by credit card 3. Time based credits, when zero time call is ended (2 minutes before ending call a message is played advising

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    Hello, We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Viber/whatsapp

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    mobile app Tamat left

    i want to get a mobile app on android and ios which can support video calling and connect to my asterisk server over ip, simple video/audio call among extensions is enough.

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    ...potential customers and setup sales appointments with upper management for website design and development. We will provide the call list of companies in our local area as well as a SIP line through Telzio to make the calls. The individual who wins this job must have good internet, at least 25/3Mbps to be able to make internet phone calls. Must possess a professional

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    I have a Linux based SIP VOIP Server, by registering any SIP IP phone with this server and giving them unique number say 500,501,502.. and they works as VOIP calling system by dialing the telephone number. Now what I want to design is, instead of IP phone use Raspberry Pi as a phone and when registered with server it should work as a Telephone. for

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    Hi kingAsterisk, we are looking to create an Asterisk integration in a call center. Quick requirements: - Operator speaks with customer and forward to PBX (with parameters like call type and call ID). - Depends on call type PBX asks questions and records the customer answer per question. - after completing, PBX sends HTTP request to a remote/local service

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    We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Viber/whatsapp executables

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    Perjanjian Kerahsiaan
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    zoho's phonebridge plugin supports up to asterisk 1.4 with asterisk 1.8 partially working (only incoming calls) with modern asterisk versions (11 and up) not working at all do to (apparently ) java problem zoho's support says that those values : dialed channel, linked channel, dialing channel are NULL instead of being filled with information. we

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    necesito crear una configuracion a travez de sms en asterisk y un redireccionamiento de llamadas entrantes, las salientes a traves de ivr y un par de detalles mas

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