Dialogic 4000 SERIES Bedienungsanleitung PDF herunterladen (Seite 38)

Dialogic

4000 Media Gateway Series Reference Guide

Page 38

Reply-To

expression:

You can configure this parameter only if you selected e-phone as Peer type in the Edit SIP Peer

Configuration window.

Enter the expression that may be necessary for the e-phone server to handle the call. Normally,

this is necessary to omit the 0 (zero) for external calls and to manipulate the address so the

e-phone server is able to call back.

Reply-To format: You can configure this parameter only if you selected e-phone as Peer type in the Edit SIP Peer

Configuration window.

Enter the format that may be necessary for the e-phone server to handle the call. Normally, this

is necessary to omit the 0 (zero) for external calls and to manipulate the address so the e-phone

server is able to call back.

Alive check: If you select this option, the failover procedure is expedited, because Diva SIPcontrol does not

wait for a call time-out if a peer does not respond.

To achieve this, Diva SIPcontrol sends "pings" periodically to the peer via OPTIONS requests. If

the peer does not send a valid answer, it will be treated as "inactive" and no calls will be routed

to this peer until the peer responds to the "pings.". In this case, Diva SIPcontrol will automatically

direct calls to this peer again.

Disconnect tone

support:

If the remote side is able to provide inband tones or signals on disconnect, check here to play

those inband tones to the SIP peer instead of terminating the SIP call immediately. The SIP call

ends either by the client sending a BYE or after the Disconnect Timer of the PSTN interface ends

(normally with "Normal call clearing").

Normally this option is set only if the peer is a human talker.

Cause code

mapping inbound:

Select the cause code mapping for calls coming from this SIP peer that you configured under

Cause Code Maps

, as described on page 57.

Cause code

mapping

outbound:

Select the cause code mapping for calls to this SIP peer that you configured under Cause Code

Maps, as described on page 57.

Codec profile: Select the codec list that you configured under Codec Profiles

, as described on page 58. If you do

not select a list, an internal default list is used with the following default priority order:

1. G.711A

2. G.711u

3. G.729, if licensed*

4. GSM-FR*

5. G.726 (16, 24, 32, and 40 kbps)*

6. Comfort Noise

7. DTMF via RFC 2833/RFC 4733 (no real codec, but internally handled as codec)

In calls from SIP to the PSTN, the first codec offered by the peer that is also in the set of supported

and available codecs is selected. This can be changed by a manual configuration that is not

currently available via the Diva SIPcontrol web interface.

*For Office Communications Server 2007, Office Communications Server 2007 R2, and Lync

Server, G.729, GSM-FR, and G.726 are disabled by default.

Maximum

channels:

Specifies the number of channels that this SIP peer is able to handle at the same time. This setting

is used by Diva SIPcontrol to distribute calls in a load-balancing scenario and to avoid speech

quality degradation and/or call failures at the peer due to overload conditions.

Early media

support:

Specifies whether the peer supports early media for calls to the PSTN. For non-human callers,

this option should be disabled.

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