AudioCodes MediaPack MP-11 Series User Manual

User's Manual 528 Document #: LTRT-27045
Mediant 1000B Gateway & E-SBC
Parameter Description
Orig. Q.850 Causes
q850-causes
[CauseMapIsdn2Isdn_OrigIsdnReleaseCause]
Defines the originally received ISDN Q.850 cause
code. For example, you can enter "127" (without
apostrophes) to represent cause code 127
Interworking, Unspecified.
The valid value (cause code) is 1 to 127.
Map Q.850 Causes
q850-causes
[CauseMapIsdn2Isdn_MapIsdnReleaseCause]
Defines the ISDN Q.850 cause code to which you
want to change the originally received cause code.
For example, you can enter "16" (without
apostrophes) to represent cause code 16 Normal
Call Clearing.
The valid value (cause code) is 1 to 127.
24.8.4 Reason Header
The device supports the SIP Reason header according to RFC 3326. The Reason header
describes the disconnection cause of a call:
 Sending Reason header: If a call is disconnected from the Tel side (ISDN), the
Reason header contains the value of the received Q.850 cause in the appropriate
message (BYE/CANCEL/final failure response) and sent to the IP side. If the call is
disconnected because of a SIP reason, the Reason header is set to the appropriate
SIP response.
 Receiving Reason header: If a call is disconnected from the IP side and the SIP
message includes the Reason header, it is sent to the Tel side according to the
following logic:
• If the Reason header includes a Q.850 cause, it is sent as is.
• If the Reason header includes a SIP response:
♦ If the message is a final response, the response status code is translated to
Q.850 format and passed to ISDN.
♦ If the message isn’t a final response, it is translated to a Q.850 cause.
• When the Reason header is received twice (i.e., SIP Reason and Q.850), the
Q.850 takes precedence over the SIP reason and is sent to the Tel side.
24.8.5 Mapping PSTN Release Cause to SIP Response
The device's FXO interface interoperates between the SIP network and the PSTN/PBX.
This interoperability includes the mapping of PSTN/PBX Call Progress tones to SIP 4xx or
5xx responses for IP-to-Tel calls. The converse is also true - for Tel-to-IP calls, the SIP 4xx
or 5xx responses are mapped to tones played to the PSTN/PBX.
When establishing an IP-to-Tel call, the following rules are applied:
 If the remote party (PSTN/PBX) is busy and the FXO device detects a busy tone, it
sends a SIP 486 Busy response to IP. If it detects a reorder tone, it sends a SIP 404
Not Found (no route to destination) to IP. In both cases the call is released. Note that
if the 'Disconnect Call on Busy Tone Detection' parameter is set to Disable, the FXO
device ignores the detection of busy and reorder tones and does not release the call.
 For all other FXS/FXO release types such as:
• no free channels in the Trunk Group,
• an appropriate call routing rule to a Trunk Group doesn’t exist, or
• the phone number isn’t found

AudioCodes MediaPack MP-11 Series Specifications

General

Brand	AudioCodes
Model	MediaPack MP-11 Series
Category	Gateway
Language	English