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    17,916 code project sip client asterisk bid 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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    ...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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    This current project 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

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    After login, receive message "cannot connect to asterisk"

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    a2billing script -- 2 6 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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    ...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 opensource too...

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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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    hi i want to register sip to sip and pass call g729 problem is i dont want to use big pc and lots of HW so i found a program its allow to do that i want [log masuk untuk melihat URL] i want to run this program on a router Router Link [log masuk untuk melihat URL] CPU: MediaTek MT7628N CPU Cores: 1 CPU MHz: 580

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    Programming Bandwith.com 5 hari left
    DISAHKAN

    Help programming sip endpoints, applications voice & sms in [log masuk untuk melihat URL] I need to connect my [log masuk untuk melihat URL] account to my PBX, be able to forward DID numbers to another number, and direct sms's messages. I would also like to complete the project described here [log masuk untuk melihat URL] Prior experience

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    im looking for asterisk pjsip developer for support our system

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    Multichannel Phone Dialer 5 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 5 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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    I need help with setting u...setting up the Outbound Parameters of my 3CX-based SIP Trunk (in the cloud). I am able to make & receive calls using Nexmo. But with Anveo, I only succeeded one time to make a call by tweaking the Outbound Parameters in the 3CX SIP Trunk Setup and I was never able to reproduce that again. Knowledge of SIP trunking a MUST.

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    sip to Native dialer APP 4 hari left
    DISAHKAN

    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for this

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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 4 hari left
    DISAHKAN

    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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    ~WE ARE LOOKING FOR ONLY NATIVE SPEAKERS~ ~THIS IS A LONG TERM FULL TIME POSITION~ This position is for [log masuk untuk melihat URL] Title:Senior Sales Executive Job Description: What will I be doing? As a T.O. Sales Closer you would be responsible for executing your position's responsibilities in alignment with our Spirit of Service culture and driving company success through perfor...

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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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    ...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 confirm transactions through SMS, as well as provide link/access

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    I want my IP PBX to be connected to Twilio WebRTC so that can be able make calls using Twilio SIP account. Calls from my PBX will come from an IP so I need someone who can make the set up for me to start using Twilio voice trunk.

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    SIP Proxy Server install 3 hari left
    DISAHKAN

    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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    you have to install skype for business 2019 and set it up. we have multiple sip phone numbers which have to be also connected to skype for business 2019 for in and outgoing calls.

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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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    Asterisk Web development -- 2 2 hari left
    DISAHKAN

    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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    Secure Issabel PBX (asterisk) 1 hari left
    DISAHKAN

    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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    Trophy icon Création logo Startup 1 hari left

    I have a project in telephony on IP for companies. It's a startup centred on cloud technologies. (Voip, Sip) The name is " Caraïbes VOIP " The main color must be blue like the sky. Je démarre une startup dans la téléphonie sur IP à destinations des entreprises. Le projet est axé sur les nouvelles technologies CLOUD. (Voip...

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    Dijamin
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    ...up/administrate the call center 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

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    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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    Hi, I deliver VoIP services to my customers and i use 90% of the time the Yealink SIP phones. The problem in 50% of the cases is that when the client is turning on their voicemail, they can't see it anyway if it is turned on yes or no. I know it is possible to use something with XML and a webserver to create a connection with the phone which can enable

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    Android application (personal purpose- no need to publish in playstore) Purpose- 1.i have more than 10000 contacts- which i cant keep in phone contacts as it affects the performance 2. and i can not give all the numbers to my personal assistant who is a telecaller because of data safety 3.i want my telecaller to know only when he communicate with a number or search for a perticular number Ap...

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    I am looking for some one who can be very good on amazon work and has did lots of work on it. Its bit trick work so please do not bid if you are not expert.

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    required a AVAYA Call Exchange PHP scripts for SIP, call recording, call records and so on ... along with DB

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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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    I need someone to setup my softphone with vpn and connect to an existing hosted pbx. my sip and voip are blocked by my isp

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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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    ...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 encrypted) - Cryptography key exchange per user (due to little time the key generation can be done as next project) - Ready and testing You

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    ...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 encrypted) - Cryptography key exchange per user (due to little time the key generation can be done as next project) - Ready and testing You

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    We are looking for Java developer who can develop C...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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    ...-assistant-using-official-docker-images/42860 . (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 on my cpanel/whm and i want to be able to see unique and realtime visitor

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    Trophy icon Asterixx Entertainment new logo 3 hari 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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    I would like to hire 2D Game Art designer for our existing game. You need to create 2D artwork along with UI. Send your portfolio and we will move from there! Thanks!

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    now i have one like when a calls come to server [log masuk untuk melihat URL] [macro-answer] exten => s,1,AGI([log masuk untuk melihat URL],${ARG1}) and php agi write all incoming calls to database an...all calls on my web panel real time my current one i can do but its take too much time to load have a look on attach file you will see my running one if you can do it please bid

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