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    11,742 freepbx asterisk 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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    CentOS/FreePBX Network Edit 6 hari left
    DISAHKAN

    I have local FreePBX server with 2 Network Interface Cards eth0: for local network DEVICE="eth0" BOOTPROTO="static" HWADDR="xx:xx:xx:xx:xx:xx" ONBOOT="yes" TYPE="Ethernet" IPADDR="[log masuk untuk melihat URL]" NETMASK="[log masuk untuk melihat URL]" NOZEROCONF="yes" BROADCAST="[log masuk untuk melihat URL]...

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    ...if I try to solve the pages does not update correctly the Vicidial/Asterisk MySQL tables. I need a correctly full working installation, the result should be an installable and fully working ISO or Virtual Machine harddisk with the working installation. Requires: dkms-dahdi, asterisk, vicidial, php7, MySQL, MariaDB, CentOS, and other of the same kind

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    asterNET modify sip trunk 6 hari left
    DISAHKAN

    asterNET modify sip trunk need a vb.net or c# to modify sip trunk in asterisk from a windows base software

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    We need to integrate an asterisk pbx with SuiteCRM if possible with freepbx or raspbx Integration means: - On inbound call open CRM calling contact information - Clicktocall from CRM - Logg inbound and outbound calls in CRM

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    Do you have interest in conference call service ? Like [log masuk untuk melihat URL] or [log masuk untuk melihat URL] ? Who can make the best portal-experience website including all features of the FreePbx or Kolmisoft platform or mor-platform ? https://wiki.freepbx.org/display/FPG/Conference+Module+User+Guide or [log masuk untuk melihat URL] So in the end we provide

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    Hello All, I'm looking an experienced Microsoft system admin. The freel...windows server 2016 needs prior to be prepared before the installation of Exchange server 2016. UC must be configured as well since we will also be hosting VoiceMail for an Asterisk VoIP system Note: Apply on this job only if you already perform such as job. Regards, Léon

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    chan_sip.c:10880 process_sdp: No compatible codecs, not accepting this offer! ... compatible codecs, not accepting this offer! I am redirecting my calls (old provider) to my new freepbx, I have a trunk added but there is an error .

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    We need someone that could write an interface to make sure that Asterisk and a Hotel Management software, Opera (that supports FIAS Protocol) can communicate, especially the billing of the calls, wakeup call, caller id.

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    ...following project: Hi, I read your gig and would like to share with you what kind of project we have in my company in Bulgaria. I am a little bit confused about PBX, VoIP, Asterisk therefor I will describe what we need as functionality. Right now in the company there are 3 hard phones, one landline number 00359 (0) 52 630 540 and one fax number 00359

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    Create a Survey FreePBX/Asterisk module to measure customer satisfaction levels for our call center agents services. Current software versions: Elastix 4 Here is what is needed. 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 1 or more fixed questions. We

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    I have a personal asterisk server for my self and kids and I see lots of messages like this. --STATUS: HANGUP on h from from ,,XXXXXX") in new stack Looks like some one trying to make calls but I dont have outgoing plans just a few extensions. I have fail2ban installed but it doesnt seem to be catching them.

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    ...fields ... the only field I have doubts about is the one labeled as follows: The difference between your server time and the [log masuk untuk melihat URL] server time (in seconds)*: The asterisk (*) at the end of the field name corresponds to the following: * Please ensure your system clock is set to proper time and time zone. AuthorizeNet gateway will validate the

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    Freepbx and A2billing support 15 jam left
    DISAHKAN

    We need support on freepbx and a2billing.

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    I need help to set-up 1 ringgroup to 1 external number. This ringgroup should then use a predefined trunk to make the outgoing call. So when I call the ringgroup I should be connected to the external number using the predefined trunk. This should also work when using inbound routes to a ringgroup. I need your remote help. Please give me a fixed price.

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    Hi! I need help with Freepbx CRM and asterisk API's: I need customers to be able to make a call from the CRM on the browser with a click on a button using an API, not with Zulu software or plug in.

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    I have asterisk server with whitelist number and make any number hit my asterisk to pass on this whitelist and if number in whiltelist it pass if not it hangup . now i need to add some options for non whitelist numbers like playback press 1 to continue and if he press add it to my whiltelist also need to send 503 code when call hangup after not passing

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    we need Asterisk Call Center System developer to work with us project base .. with full experience of : - Basic Call System - Call Center System - Call Rating System - Call Details Reports - Branding The system for us .. - CRM Integration

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    I need the full Amazon AMI Install with Freepbx tweaking in order to make this 100% functional. [log masuk untuk melihat URL]

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    Trophy icon WebRTC help needed Tamat left

    I use [log masuk untuk melihat URL] v0.11.6 for my WebRTC project. Everything works well but there is no sound on reciever side. Only caller can hear. Asterisk 16 is installed on Centos 7. Need help to fix the issue.

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    Dijamin
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    We have 3 issues with this integration: 1- For the outgoing calls, the popup shows the trunk ID number instead of the dialed number. 2- When rece...When press on the call icon in the web browser, the popup show up but the extension is not calling. Project is on high priority, Please only bid if you have experience in Asterisk with Zoho Integration.

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    ...see: The numbers of fields that has to sign on the bottom (fixed) Only the fields that has to sign with a yellow background Red borders on the bottom and an asterisk if required The text will fit inside the cell if longer will be resized. If the user use the desktop, can sign using the mouse. When the user click the save button, the

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    connect to a device grandstream voice ip phone + add sip trunk (outbound minutes) , enable recording all calls and access to them freepbx is mounted , it is working to make inbound calls and outbound calls (with softphone zoiper)

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    i have did number i want to configure in my pbx ( [log masuk untuk melihat URL] ) this is the service provider thos does this they will win the award best of luck guys /:

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    freepbx setting my number is not working

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    I have an asterisk server: Asterisk 1.8.20.0 built by palosanto @ [log masuk untuk melihat URL] I need a php based script which will popup a browser window for every incoming call & show information based on QUEUE & DNID. Budget: $50

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    want to set up my very own SIP server /VoIP system with a VPS server i will buy for the project....want to dial 800 type numbers thru MicroSIP, no web interface is needed. I need someone to fully set up the system and explain everything to me. Specifications will be sent when project is accepted.

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    IVR call flow with voice over and at least 5 API function calls. The API is SOAP 1.2. The partial IVR call flow has been uploaded to give a better understanding of what needs doing.

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    need to edit the [log masuk untuk melihat URL] from vb.net or c# urgent

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    Need someone to set us up with a Virtual PBX or cloud call center. We want to see calls in real-time and put callers in queue. Thank you!

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    Need someone to set us up with a Virtual PBX or cloud call center. We want to see calls in real-time and put callers in queue. Thank you!

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    We are facing issue where Caller did not hear voice from agent. This issue happened random time and mostly occur in the morning (Malaysia). below the detail: Version: Asterisk 1.8 & Centos 5.9 IPPBX server [log masuk untuk melihat URL] Digium Gateway [log masuk untuk melihat URL] And we have wireshark capture. PABX sent 3 SDP traffic to the phone. According to the pcap, the Digium

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    I have some trunks (inbound call numbers) I need to know which ones to call, call pop up should show the trunk name that the client called. Vtiger 6.5 or 7

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    Hola Anton F., I required Asterix Freepbx support, can you work with us.?

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    Part time support work for existing Asterisk customers, customization of recordings.

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    ...project involves migration of existing Asterisk PBX to a Azure cloud. There is existing Azure account. Install FreePBX on Azure cloud. Install Linux and FreePBX GUI. After the installation, migrate exiting asterisk configuration and voice prompts and confirm that the system works accordingly. Copies of the Asterisk files will be provided (have to be specifie...

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    I have installed and configured a system with VITALPBX, the extensions work but they are not able to configure the external line. I'm looking for someone who helps me and teaches how to configure it

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    Hello all. We need to connect 2 databases between an asterisk and a softswitch. Thanks. Fredy.

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    Configure FreePBX 14 on Vultr VPS. Set up Endpoint manager, configure firewall and help with phone provisioning

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    ...Professional Windows 10 Enterprise Windows Server 2012 R2 Windows Server 2016 Peplink Router Balance Series Pepwave SOHO series Pepwave Access Points Proxmox cluster Centos OS asterisk FreePBX Arista Switches Nortel/Avaya Switches FreeNAS Cannon MFC8080cw printers Xerox 6605n printers Tripplite UPS with SNMPWEB card installed APC Automatic Transfer Switch AP7750

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    I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

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    Hi, I have fully installed Asterisk 16 and working webphone based on sipjs. Audio on that setup are working, what I need is a someone who reconfigure it and make for me working videochat on this. It could be sipjs or sipml5 i dont care. It have to works audio call _AND_ video call

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    integracion freepbx server with didww

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    ...WEBSOCKET to work with Asterisk. (ej, mizu, duobango, kurento) 3- Tell me clearly the additional costs in scripts or licenses. 4-Demonstrate and make an example of a WEBrtc client in order to see the functioning of the client web server the voice communication. I need to know how to implement voice communication between the asterisk server and web pages

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    VAT natural language por asterisk

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    Ho la necessita di registrare un telefono cisco cp-6945 su un centralino grandstream ucm6204. il telefono ha già caricato il fw sip. Il telefono non permette la configurazione tramite web access ma solo tramite file XML

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    I need to develop an Android Studio application that allows to include the SIP / VOIP SDK to connect to ASTERISK The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Configure (add / modify / delete) SIP accounts 5.- Auto Answer Caller More details (ask) ////////////////////////////////// Necesito

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    voip freepbx Tamat left

    i need guy who has core kownledge in freepbx

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    I need dashboard to display all asterisk extensions status the same as FOP2. Clients login to managment portal and should get the specific organization extensions status including: 1. extensions status (idle,ringing,busy) 2. Idle time & talking time 3. Organization calls & queue statistics. In addition, at the managment portal agent can be log in/out

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