Pstn Fallback; Sip Call Routing Examples; Sip Call Flow Example - AudioCodes Mediant 2000 User Manual

SIP User's Manual

18.7.3 PSTN Fallback

The PSTN Fallback feature enables the device to redirect PSTN originated calls back to
the legacy PSTN network if a destination IP route is unsuitable (disallowed) for voice traffic
at a specific time. To enable PSTN fallback, assign the device's IP address as an
alternative route to the desired prefixes. Note that calls (now referred to as IP-to-Tel calls)
can be re-routed to a specific Trunk Group using the Routing parameters (see ''Configuring
iptotelrouteM2K>'' on page 264).
18.8

SIP Call Routing Examples

18.8.1 SIP Call Flow Example

The SIP call flow (shown in the following figure), describes SIP messages exchanged
between two devices during a basic call. In this call flow example, device (10.8.201.158)
with phone number '6000' dials device (10.8.201.161) with phone number '2000'.

F1 INVITE (10.8.201.108 >> 10.8.201.161):
INVITE sip:2000@10.8.201.161;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.108;branch=z9hG4bKacsiJkDGd
From: <sip:6000@10.8.201.108>;tag=1c5354
To: <sip:2000@10.8.201.161>
Call-ID: 534366556655skKw-6000--2000@10.8.201.108
CSeq: 18153 INVITE
Contact: <sip:8000@10.8.201.108;user=phone>
Version 6.4
Figure 18-24: SIP Call Flow
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18. GW and IP to IP
November 2011