Debugging - Avaya J100 Series Installing And Administering

Configuring the phone using web interface
Name
SIP Signaling Port
(minimum)
SIP Signaling Port (range)
Miscellaneous
Conference Factory URI
Voice Mail Access Code
100rel
Validate Incoming
messages
Related links
Configuring SIP settings

Debugging

Procedure
1. Log in to the web interface as an administrator.
2. In the navigation pane, click Debugging.
August 2018 Installing and Administering Avaya J100 series IP Phones in third-party call control
Description
• RTCP
Valid values are from 32 to 64,511.
The default value is 40.
Specifies the lower limit of a port range to be used for SIP signaling.
Valid values are from 1024 to 65,003.
The default value is 5060.
Specifies the port range to be used for SIP signaling.
Valid values are from 32 to 64,511.
The default value is 32.
Specifies the URI for network conferencing in third-party call control
environments.
The value contains 0 to 255 ASCII characters.
Specifies the number to access the voice mail in a non-Avaya environment.
Specifies whether the 100rel option tag is included in the SIP INVITE
header field.
The options are:
• Disable: The tag is not included.
• Enable: The tag is included.
Specifies whether AOR received in Request-URI of an incoming call must
be validated with the contact header published by phone during
registration.
The options are:
• Disable
• Enable
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