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--- sip:sip_pbx [2011/09/05 14:17]
tibor created
+++ sip:sip_pbx [2012/01/24 09:47] (current)
martinga added info about conference rooms and test agent
@@ Line 13: / Line 13: @@
 [[sip:trunk]] SIP trunks are used to connect to Internet Telephony Service Providers (ITSP) using SIP over Internet.
-===== Phone port =====
+===== Phone ports =====
-The [[sip:telephony_ports|LINE port]] of the Internet Gate is a generic FXO port that can be configured to allow all users to access the old-style PSTN global telephone network.
+The [[sip:telephony_ports|Telephony ports]] of the Internet Gate provide one generic FXO port that can be configured to allow all users access to the old-style PSTN global telephone network, and FXS ports for connecting analog telephones and fax devices.
 ===== Voicemail =====
@@ Line 20: / Line 20: @@
 ===== Auto attendant =====
-The [[sip:Auto attendant]] can answer incoming calls and allow callers make choices using their phones' keypads.
+The [[sip:auto_attendant|Auto attendant]] can answer incoming calls and allows callers to make choices using their phones' keypads.
+===== Conference rooms =====
+The [[sip:conference_rooms|Conference rooms]] provide audio conference server functionality  for up to 20 conference pariticipants. Support for PIN codes as access control, time-based scheduling of conference rooms, automatic invitation-calls to participants at conference start time, and invitiation-calls to new conference members during already ongoing conferences.
+===== Test Agent =====
+The [[sip:test_agent|Test Agent]] allows authorized SIP users to call them, and optimize the microphone and headset settings of their sip clients, by recording a message which then gets played back to the caller.

sip/sip_pbx.1315225054.txt.gz · Last modified: 2011/09/05 14:17 by tibor

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