v2.2.6-mag1000-iag801-normal-iac-v1-squashfs-sdcard-ee6dcd12.img *Added support for chinese voice v2.2.6-mag1000-iag801-normal-iac-v1-squashfs-sdcard-06385ec1.img *Fixed the problem of slow firmware upgrade for MAG1100 function boards *Fixed the problem of non-compliance of domain name detection for server address. *Fixed the problem that the SIP account is displayed incorrectly in the MAG1100 interface status page. *Fixed the problem that the MAG1000 88601 has an abnormal off-hook tone. *Fixed the problem of incorrect partition size in the latest firmware of iac-v20. *Fixed the problem of invalid first dialing timeout and inter-bit dialing timeout parameters. *Fixed the problem that Asterisk's internal time zone analysis error causes the internal time to be messed up. *Fixed the problem of recording processing program in IXU. *Fixed the problem that the IP alias of the lan port should not be displayed in bridge mode. *Fixed the problem of memory leakage caused by modifying http port on web. *Fixed the problem of batch setting of call settings in FXS port setting page. *Fixed the problem that the called party still rings after initiating call forwarding and tapping the fork again to cancel it. *Fixed the problem that cloud management sometimes does not show up *Added network configuration compatibility for dual-NIC devices. *Added support for IPV6 *Added TR069 *Added real-time printout and download of call-related logs *Added support for IPoE *Added FXS support for group configurability *Added from_domain support for IPV6 for asterisk dialog and no-dialog messages. *Added firmware upgrade support for remote firmware download. *Added support for MINDTMFDuration parameter. *Removed gateway required restriction on wan port *Optimized channel recording *Optimized OIAD binding cloud management interface *Optimized CDR page display status *Optimizedsubnet mask integer to IP string function adjustment. v2.2.5-MAG1000-iAG801-iac-v1-squashfs-sdcard-983f019f.img *Added Layer2 QoS Support *Added Pulse dialing improvements *Added asterisk codec amr *Added SIP compatibility parameter *Added line test function in web *Added SIP compatible parameter *Added option responding *Added customized config file name of auto provision *Added DTMF related parameters *Added DTMF related parameters *Added sorting policy in FXO port group *Added support for early media on s-port gateway. *Added i18n language menu options *Removed gateway required restriction in mgt settings. *Optimized the display of garbled code on the phone. *Fixed the problem that combined device does not re-register after ip change *Fixed the problem that SNMP returns incorrect port status. *Fixed the problem that CDR port lookup with duplicate channels displayed *Fixed the problem that the display mode of local number is wrong. *Fixed the problem that network jump is not reasonable. *Fixed the problem that combined platform multiple interfaces listening on 0.0.0.0 issue. *Fixed the problem that configuration send timing is not correct when auto provision works with brush retention configuration *Fixed the problem that management port configuration is cleared after closing the management port *Fixed the problem that FXS to Zibo recording card, the recording card can not record normally.*Fixed the problem that network jump is not reasonable. *Fixed the problem that Syntax error in net_mgt's get function causes logread to keep alerting. iac-v1-squashfs-sdcard-v2.2.4.img *Add Layer2 QoS Support *Add function that cloud management xfrpc ssh port down interface *Add function that mac address information to interface board information. *Add SIP compatibility parameter *Add domain name resolution configuration to page *Removed gateway required restriction in mgt settings. *Optimized Codec *Optimized firmware upgrade page *Optimized routing option *Fixed the problem that static routing affects network status page display *Fixed the problem that analog phone display time is not correct *Fixed the problem that Stability calls are filling up the CPU. *Fixed the problem that The display mode of local number is wrong. *Fixed the problem that call to SIP phone from analog phone with abnormal sound when using codec ilbc.