- #FREESWITCH AUTOANSWER ARCHIVE#
- #FREESWITCH AUTOANSWER FULL#
- #FREESWITCH AUTOANSWER VERIFICATION#
- #FREESWITCH AUTOANSWER PASSWORD#
- #FREESWITCH AUTOANSWER SERIES#
See Line Keys for examples for a list of available features.Early media is possible with Asterisk, but only in certain situations, and only with the cooperation/support of all the devices and services involved. There must be at least one line key of type 9 defined. Line keys are used for phone lines, BLF speed dials, service URLs etc. Include call statistics (packets sent and received, jitter) in SIP BYE message.ĭefines the line keys on the phone, you can specify as many lines as your phone has line keys.
#FREESWITCH AUTOANSWER SERIES#
Note: the 7900 series does not support spaces in the label. Label appears in the top of the phone's screen. Otherwise, the first line that has a message waiting is used. Incoming calls on any other lines would not be automatically answered.Īlways select the primary line when the messages button is pressed while on-hook. Prevent automatically answering of a incoming call if the phone already has a call on any line.Īllow transfers to be completed by placing the handset back on-hook.Įnable support for Voice Activity Detection (also know as Silence Suppression).Īlways select the primary line when the phone is taken off-hook. Seconds to wait before automatically answering the call for lines with set to 1.
![freeswitch autoanswer freeswitch autoanswer](https://53.cdn.ekm.net/ekmps/shops/itinstock/images/2x-jk-audio-autohybrid-analog-auto-answer-telephone-audio-line-hybrid-[3]-75609-p.jpg)
Send and receive the SIP Remote-Party-ID header, allows the called or calling party information to be updated by the $ function.
#FREESWITCH AUTOANSWER FULL#
Whether to display the full SIP URI or just the user part only. Depending on the phone model, this setting is parsed differently.Įnable remote call control. Actual phone details will be included the Remote-Party-ID header. Whether the phone hides the outgoing caller ID. Depending on the phone model, this setting is parsed differently. Whether the phone rejects incoming anonymous calls. Have the phone ring if a user hangs up an while the phone has another call on hold.Īllow transfers to be completed before the remote party has answered. Register with Asterisk, must be set to true. Up to 10 hashes can be specified by including.
#FREESWITCH AUTOANSWER ARCHIVE#
See VPN Connection for an archive containing scripts for managing certificates. Note: the path must be /svc when using the patched version of OpenConnect Server.Įncoded hash of the VPN server certificate generated by certhash. Up to 3 URLs can be specified by including. URL of the VPN server using the format HOST/svc. If it does not receive a response it will automatically connect to the VPN server.Ĭompare the host name part of the URL set below to the name in the certificate that the VPN server uses and abort the connection if the do not match. When the phone starts it will attempt to ping the TFTP server.
#FREESWITCH AUTOANSWER PASSWORD#
If disabled the user has to re-enter the password if the phone is reset. Locally store the password entered by the user. Method the phone uses to authenticate to the VPN server.
#FREESWITCH AUTOANSWER VERIFICATION#
Host name or IP address of the server running the Trust Verification Service. Port to connect to on the server running the Trust Verification Service. Up to 5 members can be defined with the phone will automatically failing-over to a member with a higher priority when it cannot connect to a member. Specifies the host name or IP address and port of the server running the Trust Verification Service. Host name or IP address of the server running Asterisk.
![freeswitch autoanswer freeswitch autoanswer](https://www.expertflow.com/wp-content/uploads/expertflow-homepage-customer-journey-img-bg.png)
SIP-TLS port to connect to on the server running Asterisk. SIP port to connect to on the server running Asterisk. Up to 5 members can be defined with the phone will automatically failing-over to a member with a higher priority when it is disconnected from the current member.Ġ to 4, the connection priority of this member, lower priorities are tried first. Specifies the host name or IP address and SIP port of the server running Asterisk. Listen to broadcasts from NTP servers for the current time Query the NTP server specified in for the current time Host name or IP address of the NTP server when using unicast mode.
![freeswitch autoanswer freeswitch autoanswer](https://www.expertflow.com/wp-content/uploads/ExpertflowCustomerInteractions-2.jpg)
Display time is adjusted by the number of minutes from UTC. D and M fields can be swapped to display date in American format. Alternate date field delimiters are - (dash) and. Preferred IP address family for signalling. An sample version of this file is included in the sample tftpboot directory from HTTP Provisioning. The actual name of the file is based on the MAC address of the phone, eg:. The main configuration file for the phone.