Setup voip site using asterisk jobs

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    15,611 setup voip site using asterisk jobs found, pricing in AUD

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

    $508 (Avg Bid)
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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 [login to view 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 : [login to view URL] [login to view URL] Http://[login to view URL] You can read about it and tell me if you can do 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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    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 4 days left

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

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    ...shall be awarded to individual who can install IP based PBX alongwith open source billing module ([login to view 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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    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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    ...Tools ([login to view 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 opensource tools to reduce the cost of operation. The Article should cover brief about tools

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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 2 days left
    VERIFIED

    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 2 days left
    VERIFIED

    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 [login to view 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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    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 1 day left
    VERIFIED

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

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

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    ...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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    SIP Proxy Server install 17 hours left
    VERIFIED

    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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    I have a very serious problem with my hosted PBX...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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    ...this and other errors from appearing again. It is an advantage if you have knowledge of both hertzner and ovh, where our servers are hosted, and loadbalance and failover IP setup....

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

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    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 [login to view 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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    ...semi-synchronous send LIVE Push-To-Talk. - No-voice-when-receiving mode; when receiving live Push-To-Talk recording other users cannot use the Push-To-Talk button. - Fast delivery; using low-bandwidth live Push-To-Talk recordings / communication. - Speaker & Playback; sender and receiver must have as look-and-feel and usability the following options; 1) All

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    ...semi-synchronous send LIVE Push-To-Talk. - No-voice-when-receiving mode; when receiving live Push-To-Talk recording other users cannot use the Push-To-Talk button. - Fast delivery; using low-bandwidth live Push-To-Talk recordings / communication. - Speaker & Playback; sender and receiver must have as look-and-feel and usability the following options; 1) All

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    We are looking for Java developer who can develop CTI & IM desktop software for our asterisk based PBX. for instant messaging we are currently using Openfire server but other XMPP servers are also welcome. We already have design and initial code to start with. Details of project will be shared on request.

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    ...[login to view URL] . (i want to use this dashboard/ analytics to see the analytics of my server , i have cpanel/whm server on centos 7, i have asterisk voip server, plex media server, i want to follow the performance of them, and i have many website

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    Trophy icon Asterixx Entertainment new logo 1 day left

    ...business). The business name is "Asterixx Entertainment". I have attached my current logo, but keep in mind this is not what I am after. If you are able to incorporate an 'asterisk' into the logo that is a bonus but not necessarily going to be the winning design. This logo will be featured on business cards, future website and facebook pages/promotional

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    now i have one like when a calls come to server [login to view URL] [macro-answer] exten => s,1,AGI([login to view URL],${ARG1}) and php agi write all incoming calls to database and web php read from DB but i dont know where is prob i saw some people they read active call every second same like console CLI real-time i need something realtime like it have to show all my active dial and all c...

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    I need someone who is expert in hacking, also expert in networks, cyber security, VOIP, Telecommunication servers.

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    PLEASE DO NOT OFFER T...This way we will quit using Twilio (or alike) database for every single call and use the one on VoipSwitch. it will not ask Twilio for the same number if it is already asked before and already the database. P.S.; definitely we do not want to use the asterisk. We would like to run it on VoipSwitch. please do not offer asterisk.

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