Asterisk openser voippekerjaan

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    15,648 asterisk openser voip 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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    Hi Vladimir U., I noticed your profile and would like to offer you my project. We can discuss any details over chat. I need to move our Asterisk system to FreePBX on Microsoft Azure cloud. I'd prefer to install it on Centos 7. I also will need to create couple extensions and connect them to a test account. There is also some basic ACD function: Dial

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    anyone who has worked with asterisk or a2 billing before please bid and i will give details of project

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    5 bida

    I need you to configure asterisk to receive calls and play an audio. Also to make calls. I have already setup asterisk and have a configuration just need to be sure

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    7 bida

    Need a back end programmer for mobile VoIP app, should have a relevant knowledge, have done similar app is a plus, knowledge in FreeSwitch, Twilio, Nexmo and Other Voip related APIs

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    Network and VoIP 5 hari left

    Hello Everyone. With more than 10.6 years’ experience in IT infrastructure management have played many important role as a Network and VoIP Service Delivery, Network and VoIP Engineer, Team Leader, End user Service Delivery, Desktop engineer with increased professional carrier and key role in IT, where have managed various line of businesses which

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    7 bida

    i need a voip build that works in apple and google with a download app for our web

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    ...Drop-down menu from which to choose the customer - Drop-down menu in which to indicate the name of the practice performed - Time taken - Small description (field with not required asterisk) - Self-compilation of the data in which it was made 7) Drop-down menu from which to choose the Cost of practice ADMIN 1) CRUD company 2) CRUD user 3) Dashboard with reports

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    Hi Arshad, we're looking for someone skilled in FreeSwitch or Asterisk to build a ringless voicemail system. We operate [log masuk untuk melihat URL] which is a popular ringless voicemail platform in the US. However, we use other provider's APIs to do the delivery. This isn't cost effective for us. We need to operate our own delivery infrastructure. Is this something

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    We are looking for a technical person who has good telecom experience and can help us developing ringless voicemail broadcasting system. There are very few companies are doing this and we have no idea how do they do that. Here is detail about companies that provide ringless VM : [log masuk untuk melihat URL] [log masuk untuk melihat URL] Http://[log masuk untuk melihat URL] You can read about it...

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    I need expert for Voice API & SDK | VoIP Calling In Your Apps | Sinch and Java Script only.

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    2 bida

    I need a modulation algorithm, which can modulate any binary bit steam, to let the bit steam go through VOIP compression and decompression, and then be recovered with minimum distortion. VOIP for example is skype, or wechat, etc.

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    ...OVERVIEW I have to shift my existing asterisk vicidial server files ( my dial plans, modules , database ,IVR , custom code on the server, user details, ALL CONFIGURATIONS. etc ) to a better hardware /server with Xeon processor 3.2 GHz.& 16GB RAM NEW VICIDIAL server details & its configuration ● Centos7, Asterisk, FreePBX & VTiger is already Installed

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    Mobile VoIP app 3 hari left

    A mobile VoIP app solution to make calls from App to phone numbers all over the world

    $805 (Avg Bid)
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    35 bida

    ...shall be awarded to individual who can install IP based PBX alongwith open source billing module ([log masuk untuk melihat URL]). The IP PbX shall be asterisk based on linux platform. VPS for installation will be provided. The payment terms include 50% payout at the time of 50% work completion and rest at the project closure. Please

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    After login, receive message "cannot connect to asterisk"

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    a2billing script -- 2 2 hari left
    DISAHKAN

    Dear all, I am seeking someone who can write a script that will install LAMP, asterisk, a2billing 2.1.1, and FreePBX automatically on CENTOS 7 by firing one or 2 command line(s).

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    Hi, I look for Freelancer who can implement Asterisk with my MGCP provider. Not SIP. Can somebody do? Thanks

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    I have requirement of asterisk PBX developer

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    5 bida

    ...Inventory/Asset Management Tools ([log masuk untuk melihat URL] with GLPI) e. Visitor Management Tools ([log masuk untuk melihat URL]) f. Printer Management Tools g. ID Management Tools (Open LDAP) h. VoIP Tools (Asterisk, FreePBX, FreeSwitch) i. Messenger Tools (Jabber) Basically the idea is to help new IT Managers and IT administrators to manage IT infrastructure using opensou...

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    Hi -I want to install freepbx and configure termination gateways on my server using vpn. I already installed asterisk and a2Billing -configure a connection with clients

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    ...MB: 8 RAM MB: 64 server send call to router >>router send to voip gateway we install our program on router and our router OS is openwrt like server to router resisted and router to Voip gateway all connection By SIP asterisk server to creytiv/re router then creytiv/re to VOIP gateway some one already made it they can run more then 64calls but

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    im looking for asterisk pjsip developer for support our system

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    Multichannel Phone Dialer 1 hari left
    DISAHKAN

    In need of software or asterisk/freepbx hosted solution for "Voice Broadcast" or "Robo Dialer". - Connects to SIP Trunk - Alternate Caller ID; Area Code Caller ID campaigns with alternating numbers - Ability to play and add audio files messages that are randomized - Multichannel lines (100) to make hundred or thousands calls per minute - Choose number

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    Asterisk 16 Queue Call Back 1 hari left
    DISAHKAN

    We have 2 platforms: 1. Freepbx 2.11 and Asterisk 11 2. FreeBpx 14 and Asterisk 16 CallBack makes callbacks simple. When your agent receives a callback request, they will see all of the caller’s information in the screen. They can click one button to accept the callback request and [log masuk untuk melihat URL] will automatically connect them to the caller. This process

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    need someone to fix the extension file on the asterisk script

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    13 bida

    i need to VOIP SIP to SIP call G729 optimization bandwidth SERVER A =asterisk server Server B =Openwrt ROuter server A&B connected to VPN VOIP device COnnected to Openwrt ROuter we send calls Server A to VOIP device SIP to SIP codec g729 its use per calls 30KBPS we want to conpress it if you can make it under 15KB we are happy i saw too many peppole

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    VOIP, SIP Expert 9 jam left
    DISAHKAN

    I’m looking for an expert of VOIP, SIP, Hack for educational purpose

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    10 bida

    We have installed OpenAnswer, FreePBX, and Asterisk on a AWS. We are looking for someone familiar with the software to finish fully setting it up for us. This includes changing all the default passwords. After setting it up we want to upgrade to PHP 7.2, and NodeJS 8. We are also looking to remove all test files, and default configuration files that

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    I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

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    we have a java script which is connected to asterisk we need someone to help change the settings

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    18 bida

    I have a vps server centos7-64 I would like to install the latest version of the following -asterisk/freepbx/a2billing/fail2ban/vpn -a2billing load balance traffic across termination gateways -IVR filter to drop test call generator (press DTMF to continue call)

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    17 bida

    I have setup an Asterisk Cloud deployment and need some help tweaking the final steps of the setup

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    19 bida

    ...portal, generate a voucher of the customers choice, enter the customers number, then click buy. The consumer will get a SMS containing Voucher number. (SMS sent out via local Asterisk Server) Major features for the initial design include: ● A minimal UI than allowed vendors to generate a unique token for client use. (UI needs to be made modern) ● App will

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    i want SIP Proxy for my voip Dialer and voip device i have asterisk server i need to register my soft phone to my SIP server but sometime we have blockage issue so i need a SIP Proxy if you can provision then good for me check the Attach FIie if you can make it let me know

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    i had asterisk/freepbx installed on centos recently. One DID on the SIP trunk is working fine but other DID has two issues. Both DIDs are on same network and similar SIP phones 1. there is a delay of 50 sec after caller presses a digit to reach staff before phones ring 2. phones ring only once....so calls are lost freelancer who installed this for

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    Need Asterisk Web Developer, Need Application like VOIP Service, Miss Call Alert service, You can check demo like myoperator , knowlarity etc

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    9 bida

    I have a very serious problem with my hosted PBX: It is available online but I already change the default 5060 to 10... Firewall and fail2ban are already setup. Sip phones can call using dialcheap and megavoip. The problem is that hackers make calls using dialcheap and megavoip and I cannot stop it. ##, *2, Tr are already disabled

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    ...SIP accounts (with hunt groups). Of course system needs configuring Asterisk for custom DID:s, call forwarding, internal calling, voicemail, ability to activate/deactivate service at any time, etc. Iam looking for a relationship with developer who will handle full deployment of asterisk with billing,GUI , with these functions : call forwarding, voice

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    6 bida

    We are looking for a company who has 24/7 server administration. We have between 4-6 servers running at the same time, and we are looking for a new company, who can be in charge of everything concerning the servers. It is important that there is 24/7 support, and that there is system admins attached, if it is not only system admins. If every case goes through level 1 or level 2 support first, the...

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    I would like create agent survey on Issabel/Elastix. Need modification in UI as well like adding menuitem under call center reports section to show Agent survey results page within elastix/issabel UI.

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    We have developed a complete workflow management software the allows clients to use the CRM that is integrated with our VOIP phone system. It is built in asterix. We need developers to maintain and manage our asterix phone system within US timezone. This will be an ongoing project. Our website is [removed by Freelancer]

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    i need a custom voip switch like astpp which can handle above 1000CPS and CC with DID Management and recording with security

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    7 bida

    we have 2 extension files we want someone to copy the dial out routine from one to the other

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    voip portal Tamat left

    development of a end user portal for our wholesale VoIP business

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    Hi, my names Dan. Programs like Skype and Discord are great but I'm looking for a custom-made VOIP to use with friends and family, something that is more locked off. I will need the ability to make calls, group calls and or channels. I'd like to have a portal where I can manage members and they have the ability to also create an account. I hope I

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    ... [root@pbx]# fwconsole restart Asterisk not currently running Running FreePBX shutdown... UCP Node Server is not running Running FreePBX startup... Starting Asterisk... [--------------->------------] 13 secs In [log masuk untuk melihat URL] line 187: Unable to connect to Asterisk. Did it start?

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    I need voip app connected to my softswitch. if you have experience in voip app, please bid. Server is IPsmarx Breeze. Thanks.

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    ...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

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    ...Push-To-Talk button + Mute button + Speaker button. - Group can be created by users added as contact (or connected into one group in back-end). - Developer can choose between Asterisk / FreePBX -or- XMPP / OpenFire as user management. - Communication must use Over-The-Air must be always 256-AES encrypted. (All live Push-To-Talk recordings must be 256-AES

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