Sip Call Flow - 3Com V7111 - VCX 4 Channels Analog FXO Media Gateway User Manual

Download the INI file using TFTP and BootP procedures and check (viewing the RS-232
3
terminal) that there are no errors.
Pick up the phone connected to port 1 of the first 8FXS Gateway and dial 102, to the
4
phone connected to port 2 of the same Gateway. Check for progress tones in the calling
end-point and for ringing in the called end-point. Answer in the called end-point and
check for voice quality.
Dial 201 from the phone connected to port 1 of the first 8FXS Gateway; the phone
5
connected to port 1 of the second 8FXS will ring. Answer the call and check for voice
quality.

SIP Call Flow

The following call flow describes SIP messages exchanged between two 8FXS Gateways
during simple call.
Phone 6000 dials 2000, sending an INVITE message to Gateway 10.8.201.161.
1
SIP Call Flow
Figure 27
MP-108
10.8.201.158
F1 10.8.201.158 ==> 10.8.201.161
INVITE sip:6000@10.8.201.161;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.8.201.158;branch=z9hG4bKacolwbzYF
From: <sip:2000@10.8.201.158>;tag=1c3535
To: <sip:6000@10.8.201.161>
Call-ID: 2123353775377NrpL-2000--6000@10.8.201.158
CSeq: 20214 INVITE
Contact: <sip:2000@10.8.201.158;user=phone>
Supported: 100rel,em
Accept-Language: en
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO
3Com VCX V7111 VoIP SIP Gateways User Manual
INVITE F1
Ringing F2
200 OK F3
ACK F4
BYE F5
200 OK F6
INVITE
MP-108
10.8.201.161
57