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Jan 06, 2025 - 07:48 AM
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Jan 06, 2025 - 07:48 AM
We’ve provided several answers to questions related to Grandstream HT801 ATA , see some other helpful Q&A related to this product.
Question: Is there a built-in router?
Answer: No, the HT801 does not have a built-in router. The absence of a router simplifies the device, focusing solely on converting analog signals to IP, which is ideal for users who already have existing network infrastructure.
Question: What is the UPC code?
Answer: The UPC code for the HT801 ATA is 6947273702269. This code is used for scanning and identifying the product in retail environments, ensuring accurate and efficient sales processing.
Question: What are the automated provisioning options?
Answer: Automated provisioning for the HT801 can be done through TR069, HTTP/HTTPS, and TFTP. These options facilitate easy configuration and management of the device, reducing setup time and potential errors in deployment.
Question: What voice codecs are supported?
Answer: The HT801 ATA supports voice codecs including G.711a/u, G.723.1, G.729a/b, G.726, and iLBC. These codecs are essential for compressing and decompressing voice data, ensuring clear and efficient voice communication over IP networks.
Question: Does it have any FXO ports?
Answer: The Grandstream HT801 ATA does not have any FXO ports. It features one FXS port for connecting to analog telephony equipment, which is sufficient for its intended use of IP telephony through existing analog devices.
Question: 1. Will the Grandstream HT801 provide power to a phone line? 2. Is it compatible with Asterisk?
Answer: 1. The HT801 allows an analog phone to connect to a VoIP network, it does not provide power to the phone line. The analog phone must have its own power supply if needed.
2. Yes, the Grandstream HT801 is compatible with Asterisk. Asterisk is an open-source IP PBX system that supports SIP (Session Initiation Protocol), which is the protocol used by the HT801 for VoIP communication. To use the HT801 with Asterisk, you need to configure the device with the SIP credentials provided by your Asterisk server.