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.
...on will be defined by BB1, BB2, and BB3 - We have 3 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 pfsense there are the agents Our typology ? Agent >> BroadBand >> Cloud VPN >> Cloud
So, I have a stock standard, fresh FreePBX installation (in production) I also have a ASP.NET project that has coding in it that pulls call information from the Asterisk database. The ASP app is currently being rebuilt and relaunched and so help is needed with someone in experience in all of the areas mentioned in the project title to put the pieces
...Android application make mobile phone act as GSM to voip gateway To connect with Asterisk PBX We needs to use an android cellphone as an asterisk channel/Gateway. His aim is to make phone calls using his Asterisk PBX through a cellphone running android. It aim to make phone calls using Asterisk PBX through a cellphone running android. call in
Our project consists of two parts: 1) Select the correct customer inside POS automat...receipts printed by Odoo POS, and the system then knows that this delivery man has delivered the scanned orders. We will provide an Odoo v9.0 server where we will also have installed a Freepbx asterisk server along with the OCA connector-telephony plugins installed.
Hi. we need part time pe...video sessions just like skype. this app has: audio, video, screen sharing, conferencing , ios native and android native apps. it is built on webRTC, Node.js , angular.js , asterisk, xcode, callkit, android studio. we need someone really expert in above skills. and you should have some apps ready similar to above skills.
Hello, I need make a integration from my FreePBX to PipeDrive CRM. It will be a PipeDr...integrated into pipedrive are below. [login to view URL] Freepbx is an ASTERISK based phone system. Need to know how long this will take. Must have prior experience with Asterisk based phone systems.
...gateway under DHCP network . we take a server which run in static IP . all client should send calls to server IP from his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from
I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.
System requirements Registration screen Logon screen online Users screen Server php and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure
I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.
I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be completed in order for push notific...
Hola con todos Alguien ha podido integar infusionsoft con centrales telefonicas ip como Elastix or Supermicro-Asterisk. La integración debe permitir : - Conocer duración de llamada - Grabar llamada - Origen de la llamada - Numero de llamadas por Gestor y por campaña de llamadas - Consolidación general de las llamadas Para controlar al gestor de
...Tabelle (ca. 20 Wörter) Total geht es um ca. 100 Texte in diesem Format zu verschiedenen Hunderassen. Preis also für 100 Texte angeben. Bitte starte dein Angebot mit einem Asterisk (*), um zu zeigen, dass du das Inserat gelesen hast....
i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test
We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux
...“Neverwrote”, for managing virtual notes—The backend shall run on a Node.js server, and that the frontend shall be built with React. You will only need to work in the api/ directory (for the backend API) and the frontend/ directory (for the frontend interface). The NGINX server is already correctly configured, as are all of the Docker-related things
Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined
Hello freelancer, I currently need FreePBX installed with our asterisk instance. we have a 5 hardphones, 15 softphone users. We are in need of IVR setup, conferencing, outbound/inbound routes and trunks. Our freepbx will need to be setup based on best security practice and monitored for atleast 15 business days after installation. we are moving from
i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [login to view URL] or yate or besip [login to view URL]
Hi, we need a professional Asterisk / freepbx sysadmin who can fix the nating in dmz using opnsense. currently we have done 90% of work Lan 192.168.X.0/24 WAN A.B.C.D (static public) DMZ 192.168.Y.0/24 PBX is here (also VIP to L.M.N.P public IP) using 1:1 Nating now the odd thing is that we are using linphone as softphone and we are having problem
...documented installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording •...
...documents are made with LibreOffice and changed to PDF for use. The server used has 3 drives. One for the Nginx/Apache Web Server and the SMTP email Server, one for the MySQL Data Base Server and one for the module with LibreOffice. Invoice information comes from another Asterisk Server. To start, Teamviewer must be used, as some software is already ins...
I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.
**Must speak both english and spanish** We are developing an inhouse an Asterisk based PBX solution to suit our needs. This is an ongoing project and we will need developers who are experts in PHP, NodeJS, CSS, MySQL, Optimization, Security. Bonus points for mobile development. We also need project managers. Previous experience managing development
I have a FreePBX/Asterisk System working at Amazon. I can access it directly or via a VPN. Normal telephony works as expected. 1) Trying to get WebRTC phone (via the UCP) working 2) Trying to integrate external WebRTC Can you help debug?
...capability (CHECK on this) if not we will have to go open source with asterisk pbx, -Company with PBX need to be able to manage DID (phone numbers) and assign number to agents, administrator and super admin account, -Lower the cost per minute and per text (that is why we need to migrate to asterisk open source) -Lastly I need to be able to receive text message
We have an asterisk PBX integrated with Zoho CRM but it's delivering the channel ID instead of the dialed number to the the CRM extension. We need some one to fix the coding of the asterisk PBX to deliver the correct needed information
...system 53 - Insurance lead generation 54 - Inbound Call Tracking 55 - Outbound Call Tracking 56 - Call Analytics integrate Asterisk call center with CRM to have some feature bellow: - Sale can use Call feature on CRM via Asterisk. - Can use feature Click to Call? Please show me the solutions (brief, draft with your paper) to continue!...
I want to create a ARI program that will play a message from URL in cloud and record the sentence spoken in channel and repeat what the person spoke after record is done Key thing here is I do not want any button pressing...Using ARI functions code should detetc when a person has spoken and when there is silence of more than 2 secs , consider the recording done Ideally i would like to implement t...