Dialogic 4000 SERIES Manual de usuario Pagina 8

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Dialogic
®
4000 Media Gateway Series Reference Guide
Page 8
Basic call incl. numbering services:
Called Party Number
Calling Party Number
Redirecting Number
Call Routing
Call Hold/Retrieve (e.g., Re-Invite mapping towards ISDN)
SIP-side Call Transfer as transfer target (C-party) and as transferee (A-party)
PSTN-side incoming Call Diversion
Support for SIP 302 REDIRECT, which works as follows: a SIP call is redirected per 302 Redirect with new IP
port numbers (as used with Microsoft
®
Office Communications Server 2007, Microsoft
®
Office
Communications Server 2007 R2, and Microsoft
®
Exchange Server 2007 without changing the IP address)
Support for SIP Refer: a SIP call is redirected per SIP Refer to a new SIP target, using Replaces and Referred By
SIP Session Timer (RFC 4028)
Simplified Number Normalization based on PSTN connection parameters
Number Manipulation using Regular Expressions
Media Processing
Support for the following codecs:
G.711 A-law and u-law
G.726 (16, 24, 32, and 40 kbps)
•G.729*
•GSM-FR
•iLBC**
•sRTP
RTP dynamic payload audio/telephony event
RTP profile RTP/AVP
DTMF via RTP payload/telephony event (RFC 2833 or RFC 4733)
PSTN-side fax tone detection via RTP event (RFC 2833 or RFC 4733)
128 ms Echo Canceller supported on all boards, and additionally, 256 ms EC supported on those boards listed
in MultiPRI boards section.
*For G.729, you need to purchase and activate a license before you can use it. For more information, see the
Dialogic
®
4000 Media Gateway Series Quickstart Guide, which is available at
http://www.dialogic.com/manuals/dmg30004000
. G.729 is only available on Dialogic
®
Diva
®
Multiport V-PRI
Media Boards.
**iLBC is only available on Diva Multiport V-PRI Media Boards. On Dialogic
®
Diva
®
V-4PRI/E1/T1-120 PCIe HS
boards
and Dialogic
®
Diva
®
V-8PRI/E1/T1-240 PCIe FS boards, up to 18 channels for each PRI port are supported.
Reliability
Load balancing and failover on PSTN side
Load balancing and failover on SIP side (optionally uses OPTIONS for keep-alive check)
Alive check for active calls on SIP side via SIP session timer
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