Here are some essential steps that you should not overlook when preparing to launch a new business from your home.
Tengo un Vps en digital ocean (este es un FreePbx), he aumentado el el tamano de su disco duro pero no se actualiza en mi FreePbx, necesito alguien experto para ampliar la particion de mi servidor sin perder la informacion que tengo en esa particion
Hi, I am getting the source code for an ongoing project. The ASTPP switch already has API's built for the Linphone. The linphone code is pointing to an old switch. Step one: I just need the linphone to be migrated to a new switch. Step 2: 1. Bundle/Packages and DID -users can choose from —#different bundled minutes(ie. 80 mins to destination X & Y for $5, valid for Z amount of days.), ---#unlimited plans to certain destinations* —#user can buy DID (DID will show as caller ID for that user) 2. Caller ID For regulatory reasons, we can't modify the caller ID to be that of another local provider.. -when users call from the App, it need to show caller ID unavailable (until that person pays for a DID, or a bundle that includes a DID.) -Then when the ca...
Hello We need to build a new website for providing our own VOIP services to customers. We require a proffessional to set up the initial system and provide on-going support and development. Previous experience with asterisk servers, setting up Architecture, virtual pbx, sip trunks, etc' customer billing, my account, etc' The end goal is to be able to start selling voip services. We prefer a person who did something like that before. I will be happy to answer questions.
Looking for Freelancer trainer who can deliver CISCO HCI training. Need to support the HCI infra for ongoing contract with VALE for NSSR and support work CHXSE 700-905 Cisco HyperFlex for Systems Engineers Exam (Training &Certification
Vertriebsassistent/in im Home Office gesucht Für unsere Unternehmensberatung (Schwerpunkte Telekommunikation & IT) suche ich ab sofort ZUVERLÄSSIGE und ERFOLGSORIENTIERTE Vertriebsassistenten/innen Auf 450,- €-Basis, Teilzeit oder freiberuflich Bei Eignung spätere Vollzeitbeschäftigung gern gesehen Ihre Aufgaben: Terminvereinbarung bei Neu & Bestandskunden & Terminkoordination Ihr Profil: - Erfahrung in der Terminierung und der Telekommunikation & IT Branche erforderlich - Am Telefon sind Sie klar und gut verständlich und verstehen es sich auszudrücken - Sehr gute Deutschkenntnisse in Wort und Schrift Nach erfolgreicher Einarbeitung und erfolgen in der Terminierung ist auch die Übernahme von zusätzlichen Aufgabenbereichen m&o...
We get a phonenumber out of scraped api, we sent this to Twilio. In total this proces takes 10 seconds before phonecall is setup, which i want reduced as fast as possible. Preferrable instant. Anyone who can help
in voip for USA/Canada LRN/LNP query is conducted to get the shortest path and query server send SIP 302 redirect messages as required. We will update its database manually but it has to respond to query's fast and accurately as per pre-loaded data. Only knowledgeable person in voip architecture should bid to this project and i will pay only bidding price, no negotiation on price shall be done after bidding.
Hi, I need someone who has an excellent experience in the FREE Open-Source SIP SBCs to suggest the vendor we will install and help me to install it Kindly don't bid if you don't have the experience that is needed, Thanks!
We have Server of HIK VISION Installed software . We have Workstation with software we have NVR for Recording . we have 10 Cameras CCTV all are install we need final setting which will be full job . 1) Camera should record . 2) Workstation should be able to connect with server of HIKVision . 3) NVR should be able to record when needed. 4) Further different settings needed. based on the functions of HIKVision . 5) all are connected through one Switch for Testing . 6)once you do the setup you will prepare the configuration and setup document and will hand over to us . We will give you remote access of server through anydesk to setup and configure . we will need 6 Months support as well .
Looking for a Vonage API expert. I have a CRM developed in PHP. My sales team use Vonage phones to call customers. I want an option to make call and receive call from customers from the CRM without our team able to see the customer number. Vonage Provide APIs for that. I need someone who already have an experience in Vonage API integration click to call functionality. Please apply with relevant example where the same feature is implemented.
This app is controlled by super admin. Super admin can create a client main account. This client main account can approve/delete new user. The user account can do VoIP calling between the user. This user account can also add other user into their list.
Need support on Cisco Router, Switches and ASA firewall Project Help
hello we are looking for someone to create a VOIP app or website to call people in Saudia Arabia, we are looking to call numbers in saudia arabia and that app provide us calls & landlines & mobile numbers for us to make calls.
Build a voip app with dashboard for admin control reseller panel , server management Use freepbx for calling system Requirement to be discussed more in debt but mainly use sip trunk to make calls, Caller ID selection User interface for user info, credit remaining and extort dates Admin dashboard to control all user info Android based, source code to be provided too
Create custom module in prefex crm for integration with asterisk with following features: - create/edit extensions - create/edit trunks - create/edit outbound routes - create/edit inbound routes - create/edit IVR menu - calls report and recording - popup up screen when incoming calls with customer details or add new option. - click to call from crm. - conversation history in customer details. Thanks
I require assistance to set up my existing Amazon Connect instance with a voicemail solution. The solution should: - Utilize the cloud deployment template created by AWS - Be incorporated into an existing Contact Flow, and initiate a voicemail if no agents picks up incoming call or if inbound is after business hours - Send an email, using transcribed voice-to-text, including an attachment of the audio recording
I want my ASTPP Server to make outbound and Inbound calls from the Sip trunk routed from its second ethernet interface which it can primarily see when you ping directly. but says Gateway Ping Failed on Fs_cli