Why does my Yealink SIP-T58V/A IP Phone display “No Service”?
If your Yealink IP Phone displays “No Service”, ensure that: * An account is actively registered on the IP phone at Settings->Status->Accounts. * The SIP account parameters have been configured correctly.
General| Resolution | 1024 x 600 |
|---|---|
| HD Audio | Yes |
| Weight | 1.2 kg |
| Operating Humidity | 10% to 90% (non-condensing) |
| Display | 7-inch touchscreen |
| Video Resolution | 720p |
| Wi-Fi | Yes |
| Bluetooth | Yes |
| USB Ports | 2 USB ports |
| Ethernet | 2 x 10/100/1000 Mbps |
| PoE Support | Yes |
| Audio Codecs | G.711, G.722, G.729, Opus |
| Voice Features | HD Voice |
| Headset Support | Yes |
| Expandable | Yes, supports expansion modules |
| Supported Protocols | SIP |
| Operating Temperature | 0 to 40 °C |
| Operating System | Android 5.1.1 |
Information about the scope and purpose of this administrator guide for IP phones.
Overview of the chapters included in the administrator guide, detailing IP phone features and configurations.
List of related documents for SIP-T58V/A and SIP-T56A IP phones, including Quick Start Guides and User Guides.
Typographic and writing conventions used in Yealink documentation for clarity and consistency.
Guidance on understanding conventions used in summary and configuration parameter tables for effective configuration.
Explanation of the summary table format indicating three provisioning methods for feature configuration.
Details on the configuration parameter table format, explaining how to read and understand parameter entries.
References for configuring other Yealink products and accessing product support and release notes.
Basic knowledge of VoIP principles and SIP components for better understanding of phone deployment scenarios.
Explanation of VoIP technology, H.323, and SIP protocols used in voice communications over IP networks.
Description of SIP components, including User Agent Client (UAC) and User Agent Server (UAS) roles.
Introduction to SIP-T58V/A and SIP-T56A IP phone models and their role in network topology.
Detailed list of physical features for SIP-T58V/A and SIP-T56A IP phones, including screen, ports, and LEDs.
Overview of key features supported by IP phones, including phone, codecs, voice, video, network, and management features.
Introduction to EXP50 expansion modules, their functionalities, and physical features.
Requirements for IP phones to operate successfully as SIP endpoints in a network.
Explanation of how Yealink IP phones connect to networks and their capabilities beyond traditional business phones.
Step-by-step instructions for installing IP phones, including camera, stand, handset, power, and network connections.
Instructions for inserting the camera into SIP-T58V/A IP phones, specifying camera compatibility.
Methods for attaching the desk stand and the optional wall mount bracket for SIP-T58V/A and SIP-T56A IP phones.
Instructions on how to adjust the angle of the touch screen for SIP-T58V/A IP phones.
Instructions for connecting the handset and optional headset to SIP-T58V/A and SIP-T56A IP phones.
Steps for connecting AC power and network cables to SIP-T58V/A and SIP-T56A IP phones.
Instructions on connecting a USB flash drive to save captured pictures, recorded audios/videos, or manage data.
Overview of the IP phone's initialization process, including loading ROM file, configuring VLAN, querying DHCP, and updating firmware.
Steps to verify the IP phone's startup process, including checking status and readiness for use.
Configuration of network parameters for IP phones, including DHCP, IPv4/IPv6, VLAN, and 802.1X authentication.
Instructions for setting up IP phones with a provisioning server, including configuration, firmware upgrades, and personalized settings.
Explanation of the power indicator LED states and configuration options for power light behavior.
Configuration of notification popups for missed calls, incoming calls, and new voice mails.
Customizing the wallpaper displayed on the IP phone idle screen and EXP50 expansion module.
Configuration of screen saver types, wait time, and display settings for IP phones.
Configuration of power saving features, including office hours, idle timeouts, and intelligent power saving modes.
Configuration of touch screen backlight brightness and backlight time delay.
Configuration of Bluetooth mode, device name, and call permission for Bluetooth connections.
Configuration of the breathing light to indicate line key statuses, especially for multiple line keys.
Configuration of page switch key LED on expansion module for BLF monitored user incoming calls.
Procedure for registering SIP accounts, including parameters for account information and server settings.
Customizing DSS keys to be automatically assigned with Line type, associating multiple keys with an account.
Configuration of call information display methods for incoming and outgoing calls.
Customization of account information display on the pre-dialing or dialing screen.
Configuration of IP phone time and date settings, including NTP time server, time zone, and time formats.
Specification of languages for phone user interface and web user interface.
Customization of soft key layout above Android keys for different call states.
Configuration for assigning the pound key (#) or asterisk key (*) as the send key.
Definition of dial plan using regular expressions or XML template files.
Configuration of emergency dial plan for dialing emergency telephone numbers.
Configuration of hotline feature for automatically dialing a preset number after lifting handset or pressing a key.
Configuration of off hook hot line dialing feature for automatically dialing a pre-configured number.
Configuration of search source list for automatically searching entries based on entered string.
Management of local directory, including adding contacts and groups via local contact files.
Configuration of live dialpad feature for automatically dialing entered phone number without pressing send key.
Configuration of speed dial keys using dedicated DSS keys for frequently used numbers.
Configuration of call waiting feature and call waiting tone for receiving new incoming calls during an active call.
Configuration of auto redial feature for redialing busy numbers and setting attempts and waiting time.
Configuration of auto answer feature for automatically answering incoming calls and setting auto-answer delay.
Configuration of IP direct auto answer feature for automatically answering IP address calls.
Configuration of allow IP call feature for receiving or placing IP address calls.
Configuration to enable IP phones to only accept SIP messages from the SIP server and outbound proxy server.
Configuration of call completion feature for monitoring busy party and establishing call when available.
Configuration of anonymous call feature to conceal caller identity information displayed on the callee’s screen.
Configuration of anonymous call rejection feature to automatically reject incoming calls from concealed identities.
Configuration of DND feature to ignore incoming calls and log them in the Missed Calls list.
Configuration of busy tone delay to define the period of time during which the busy tone is audible.
Definition of return code and reason of SIP response message for refused calls.
Information about early media (audio and video) played to the caller before a SIP call is established.
Defines whether to deal with the 180 message received after the 183 message for call setup.
Configuration for using an outbound proxy server within a dialog for all SIP request messages.
Configuration of SIP transaction layer timers T1, T2, and T4 for managing SIP sessions.
Explanation of call hold service for placing active calls on hold and supporting different call hold methods.
Configuration for playing recorded music to fill silence when a call is placed on hold.
Configuration of call forward feature to redirect incoming calls to a third party.
Configuration of local conference for processing audio of all parties on the IP phone.
Implementation of network conference using REFER method for flexible call with multiple participants.
Configuration for allowing other parties to remain connected when the conference initiator drops the call.
Capability to synchronize feature status (DND, Call Forward) between IP phone and server.
Configuration of transfer modes (New Call, Blind Transfer, Attended Transfer) for DSS keys.
Configuration for picking up incoming calls on a specific extension using directed pickup or DPickup key.
Configuration for picking up incoming calls within a pre-defined group using group pickup or GPickup key.
Implementation of call pickup using SIP signals and INVITE message with Replaces header.
Feature allowing users to view placed calls list on pre-dialing screen for quick dialing.
Functionality to place a call back to the last caller using a recall key.
Configuration for automatically filtering designated characters when dialing numbers.
Feature allowing users to park a call on an extension and retrieve it on another phone.
Allows IP phones to display caller identity from SIP header in INVITE message.
Allows IP phones to display identity of connected party for outgoing calls from SIP header.
Description of remote phone book stored on remote server, accessible via URL.
Application protocol for accessing and maintaining information services for distributed directory over IP network.
Monitoring user status (busy or idle) on IP phones using BLF keys.
Monitoring specific extensions for status changes and receiving notifications.
Enables IP phone to display feature name instead of access code when dialing.
Allows users to share an extension registered on multiple IP phones simultaneously.
Informs users about the number of messages waiting in their mailbox without calling the mailbox.
Allows IP phones to send/receive Real-time Transport Protocol (RTP) streams for paging.
Supports recording calls by tapping the call record key, saving files to internal SD card or USB flash drive.
Allows users to clear registration configurations and register account on line 1.
Provides the logon wizard during the first startup for IP phones.
Allows IP phones to interact with web server applications by sending HTTP or HTTPS GET requests.
Allows IP phones to interact with web server applications by receiving and handling HTTP or HTTPS GET requests.
Ensures continuity of phone service by deploying two separate servers for failover and fallback purposes.
Utilizes configured domain name of the server resolved through DNS, or statically configured DNS records.
Technical specification for secure auto-configuration of Customer Premises Equipment (CPE).
Allows IP phones to continue playing dial tone after inputting preset numbers on the pre-dialing screen.
Selection of built-in system ring tones or custom ring tones for the phone or account.
Triggers distinctive ring tones for specific incoming calls based on Alert-Info header.
Customization of warning tones for various conditions like dial, busy, congestion, and voice mail.
Configuration of warning tone played when receiving a new voice mail.
Configuration of ringer device (speaker, headset, or both) for incoming calls.
Allows preferential use of headset if physically connected to the IP phone.
Enables users to use two headsets on one IP phone for full-duplex or listen-only conversations.
Adjustment of sending volume for handset, speakerphone, or headset during calls.
Information on audio codecs, supported codecs, and RTPmap for configuring audio transmission quality.
Features for reducing acoustic echo and improving voice quality, including AEC, BNS, AGC, and VAD.
Methods of transmitting DTMF digits on SIP calls: RFC 2833, INBAND, and SIP INFO.
Feature for generating quality metrics (e.g., MOS-LQ, MOS-CQ) for voice quality monitoring.
Changing default user ('user') and administrator ('admin') passwords for web interface access.
Configuration of timeout interval for automatic logout from web user interface after inactivity.
Using phone lock to prevent unauthorized use by locking keys and screen, requiring a password to unlock.
Protocol for secure communications, enabling encrypted message transmission and provisioning via HTTPS.
Encrypts RTP during VoIP calls to avoid interception and eavesdropping, requiring simultaneous enablement.
Support for downloading encrypted files and encrypting files before/when uploading them to the server.
General information for troubleshooting common problems using log files, packets, status indicators, etc.
Procedure for exporting system logs to local system, syslog server, or FTP/TFTP server.
Methods for capturing packets via web user interface or Ethernet software for analysis.
Automatic phone reboot feature when detecting a fatal failure.
Obtaining IP phone information from status indicators like power LED, line key, and on-screen icons.
Viewing video and audio data during an active call via web phone user interface.
Exporting and importing configuration files for checking and troubleshooting phone issues.
Solutions to common issues encountered while using the IP phone.
Troubleshooting steps for IP phone not getting an IP address and resolving IP conflicts.
Troubleshooting steps for incorrect time and date display, including NTP server configuration.
Troubleshooting steps for issues like blank touch screen or 'No Service' message.
Explanation of the difference between remote phone book and local phone book.
Troubleshooting steps for poor sound quality, low volume, or no sound during calls.
Troubleshooting steps for bad video quality, including display resolution and packet loss.
Troubleshooting steps for low wireless signal strength, connection issues, and Bluetooth device pairing.
Troubleshooting steps for firmware upgrade failures and verifying firmware generation and version.
Information on auto provisioning and Plug and Play (PnP) for acquiring provisioning server address.
Explanation of Plug and Play (PnP) method for IP phones to acquire provisioning server address.
Troubleshooting steps for exporting system logs to provisioning or syslog servers.
General troubleshooting for common issues and ways to reset the IP phone to factory configurations.
Glossary of terms and abbreviations used in the administrator's guide.
List of supported time zones, including their corresponding names and offsets.
List of trusted Certificate Authorities (CAs) that Yealink IP phones trust by default.
Parameters for configuring DSS keys, including line key, programmable key, and ext key.
Flowchart illustrating auto provisioning for Yealink IP phones, focusing on preserving user personalized settings.
Explanation of static settings and their differences from other parameters, including network and auto-provisioning settings.
Description of Yealink IP phones' compliance with IETF definition of SIP, including RFC and Internet Draft support.
List of supported RFCs and Internet drafts related to SIP and its extensions.
Supported SIP request messages, including REGISTER, INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, REFER, PRACK, INFO, MESSAGE, UPDATE, PUBLISH.
Supported SIP request headers, indicating which headers are sent and properly parsed.
Supported SIP responses, categorized into Provisional, Successful, Redirection, Request Failure, Server Failure, and Global Failures.
Explanation of SDP headers and their usage in SIP sessions.
Illustrations of SIP call flows, including successful call setup, unsuccessful calls, and call transfers.
Scenario of a successful call between two Yealink SIP IP phones, User A and User B.
Scenario of an unsuccessful call caused by the called user being busy.
Scenario of a successful call setup and call waiting between multiple Yealink SIP IP phones.
Call flow scenario illustrating a blind transfer of a call to a third party.
Call flow scenario illustrating an attended transfer of a call to a third party with consultation.
Call flow scenario illustrating successful call forwarding when User B is busy.
Call flow scenario illustrating successful call forwarding when User B does not answer.
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