User's Manual

SZ; Reviewed:

SPOC 8/10/2007

Solution & Interoperability Test Lab Application Notes

29 of 30

ACM-Cisco-CM-GK.doc

• Use the command status station 4555 from the SAT interface on Avaya Communication

Manager to check the status of this Avaya IP telephone. As shown from page 4 below,

this is a call using codec G.711MU from extension 4555 to S8300 Media Gateway VoIP

engine (22.1.1.1) and the connection-type is ip-tdm.

status station 4555 Page 4 of 7

AUDIO CHANNEL

Port: S00001

Switch IP IP

Port Other-end IP Addr :Port Set-end IP Addr:Port

G.711MU Audio: 1 22. 1. 1. 1 :2050 22. 1. 1.222:61974

Node Name:

Network Region: 1 1

Audio Connection Type: ip-tdm

Port: S00001 Shared Port:

Product ID and Release: IP_Phone 1.500

H.245 Tunneled in Q.931? does not apply

Registration Status: registered-authenticated

MAC Address: 00:04:0d:ec:06:c6

Native NAT Address: not applicable

ALG - NAT WAN IP address: not applicable

Authentication Type: DES-56-plus

• Page 6 shows the same call from S8300 Server VoIP engine (22.1.1.1) to a Cisco phone

(60.1.1.100) using codec g729.

status station 4555 Page 6 of 7

SRC PORT TO DEST PORT TALKPATH

src port: S00001

S00001:TX:22.1.1.222:61974/g711u/20ms

001V038:RX:22.1.1.1:2050/g711u/20ms TX:ctxID:292

001V037:RX:ctxID:292 TX:22.1.1.1:2052/g729/20ms

T00091:RX:60.1.1.100:24204/g729/20ms

• Change bandwidth to 16 kbs on Gatekeeper for remote zone (Location A). Place two

calls from Cisco CallManager to Location A and verify the fist call is successful and the

second call was rejected by gatekeeper because no more bandwidth available.

• Verify that caller ID (calling name and number) can be displayed correctly.

6. Conclusion

As illustrated in these Application Notes, calls can be made between Avaya Communication

Manager and Cisco CallManager via a Cisco 2811 Gatekeeper.