
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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