AltiGen MAXCS 7.0 Update 1 ACM Administration - Manual (Page 141)

Configuring Virtual Board SIPSP
MaxCS 7.5 Administration Manual 125
1. Connect the CSU (Adtran model T1 CSU ACE used as an example) to the T1/PRI or T1/E1 PRI board
using an RJ-48C or RJ-48X cable.
2. Connect the CSU to the network termination box using an RJ-48C or RJ-48X cable.
AltiGen T1 Socket (RJ-48)
Refer to your CSU manufacturer’s
manual for the proper pinout.
Pin 1=Receive Ring (INPUT)
Pin 2=Receive Tip (INPUT)
Pin 4=Transmit Ring (OUTPUT)
Pin 5=Transmit Tip (OUTPUT)
Note: CSUs also are used for line lengths over 75 feet, which helps to resolve attenuation issues.
Troubleshooting T1/E1 Common Symptoms
The most common problems when installing T1 CAS or T1 PRI services:
1.
The service provider misconfigures your T1 CAS/T1 PRI service or terminates your service improperly.
2. T1 is installed but not turned on because there is no termination device for a period of time.
3. T1 is turned on but channel is not in service.
MAXCS provides basic troubleshooting information in the T1 Span Configuration window, described in “T1 and
E1 Configuration” on page 114.
Configuring Virtual Board SIPSP
A VoIP connection typically consists of two parts.
• Signal Channel – Responsible for setting up and tearing down a call using protocol. For example, SIP
protocol is used in MAXCS to build a signal channel between the server and the IP phone.
• Media Path – Responsible for encoding, transmitting, and decoding voice for both parties. For example,
when an IP phone user makes a call to an outside number, the voice will be encoded at the IP phone,
transmitted to the system via the IP network, decoded by the VoIP codec, and passed to a trunk port so
that the external party will hear the voice.
The purpose of the virtual boards SIPSP is to build signal channels for different connection types, IP exten-
sions, SIP Tie Trunks, and SIP Trunking from ITSP. Each channel will have its channel ID similar to channels
on a Triton extension or trunk board. When an IP phone registers to the system, a channel ID will be assigned
to the IP extension. However, these channels are only responsible for processing protocol and call control sig-
nals. They require a media path from a VoIP board or from the IP phone to establish a voice steam so that both
sides can hear.