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Abstract
These Application Notes illustrate a sample configuration using a SIP private network for
inter-site calling among four sites. Two sites are equipped with Avaya Communication
Manager, and two are equipped with Avaya Distributed Office. Although calls among all sites
are verified, the configuration and operation of the sites running Avaya Communication
Release 5 are the focus of these Application Notes. Two new SIP-related trunk features in
Avaya Communication Manager Release 5 are illustrated. First, independent Avaya
Communication Manager Release 5 systems networked via SIP trunks can benefit from SIP
private networking enhancements that enable certain features to operate between networked
systems akin to the way the features would work within a single system. Examples include
priority calling, display enhancements for answering party display updates, and displayed
reasons for call redirection occurring at a networked site. Second, Look-Ahead Routing has
been enhanced, enabling calls that are delivered to the SIP network to be automatically
redirected to alternate routes should the SIP network fail to respond, or respond with specific
SIP response errors that are outlined in these Application Notes. The illustrated configuration
builds upon previously published Application Notes that fully describe three of the sites in the
configuration, including both Avaya Distributed Office sites.
Figure 2 shows the network used to verify these Application Notes. Figure 2 builds upon the
network shown in Figure 1 by introducing a new end-user site in the upper left quadrant of the
diagram. The Avaya SES Edge and Avaya Distributed Office Central Manager visuals from
Figure 1 are de-emphasized in Figure 2 in the upper left quadrant, and an Avaya G250-DCP
Media Gateway with Avaya S8300B Server is added. Like the other three sites, the fourth site
will be assigned a unique “Branch Prefix” (e.g., 25) to enable calls from any site to reach this
newly added site via the SIP private network. Unlike the other sites, the new site will not use
local telephones running the SIP protocol, but will use the SIP private network for inter-site
calling. Avaya Digital telephones are directly connected to the Avaya G250 Media Gateway,
and Avaya IP Telephones running the H.323 protocol register with the Avaya S8300B Server. If
the SIP private network does not respond, or responds with specific SIP errors, Look-Ahead
Routing can be used to complete outbound calls from the Avaya S8300B Server via an alternate
route. In the sample configuration, calls “look-ahead” to an analog trunk in the Avaya G250
Media Gateway.
For inter-site dialing, a caller at any location in Figure 2 may dial the Automatic Alternate
Routing (AAR) access code, a location prefix, and extension. In the sample configuration, the
AAR access code is uniformly configured to be “8”. The length of the private network location
prefix is configurable. In the sample configuration, a two-digit prefix length is used. The
prefixes for each location are indicated in Figure 2.
Alternatively, inter-site calls dialed from the Avaya Communication Manager sites may dial a
five digit Uniform Dial Plan (UDP) number. The newly introduced site in the upper left
quadrant of Figure 2 has extensions of the form 582XX. The Avaya Communication Manager
site retained from Figure 1 has extensions of the form 584XX. The two Avaya Distributed
Office sites use 3 digit extensions. The two digit prefix code followed by the three digit
extension can be dialed as if the dialed string were a five digit UDP number from either of the
Avaya Communication Manager sites. The various dialing options are shown in Table 1 and
Table 2.
In the sample configuration, all inter-site calls will complete using G.729. All connections
between IP Telephones, intra-site, and inter-site, will use IP-IP Direct Audio, such that the final
end-to-end media path will be directly between the two telephones.
Call Type Example Caller Call Recipient Number to Dial Final Result
Intra-branch 220 G.711MU
Distributed x208 x220 -- or -- “Ip-Direct”
Office i40 Site 8-40-220 connection
Intra-branch 202 G.711MU
Distributed x203 x202 -- or -- “Ip-Direct”
Office i120 Site 8-20-202 connection
Intra-branch
Avaya S8300C x58467 x58430 58430 G.722
Server Site -- or -- “Ip-Direct”
8-45-58430 connection
DO i40 Site G.729
To x220 x203 8-20-203 “Ip-Direct”
DO i120 Site connection
DO i40 Site to
Avaya S8300C x208 x58467 8-45-58467 G.729
Server Site “Ip-Direct”
connection
DO i120 Site G.729
To x203 x208 8-40-208 “Ip-Direct”
DO i40 Site connection
DO i120 Site to G.729
Avaya S8300C x202 x58467 8-45-58467 “Ip-Direct”
Server Site connection
Avaya S8300C 40208 G.729
Server Site to x58467 x208 -- or -- “Ip-Direct”
DO i40 Site 8-40-208 connection
Avaya S8300C 20203 G.729
Server Site to x58430 x203 -- or -- “Ip-Direct”
DO i120 Site 8-20-203 connection
Table 1 – Example Calls and Expected Results Repeated from Reference [1]
Call Type Example Caller Call Recipient Number to Dial Final Result
Intra-branch Digital 58203 G.711MU
S8300B Server x58220 Telephone -- or -- IP Telephone to
Site x58203 8-25-58203 Avaya G250
Media Gateway
DO i40 Site to Digital G.729
S8300B Server x208 Telephone 8-25-58203 IP Telephone to
Site x58203 Avaya G250
Media Gateway
DO i120 Site to G.729
S8300B Server x202 x58220 8-25-58220 “Ip-Direct”
Site connection
Avaya S8300C 58220 G.729
Server Site to x58467 x58220 -- or -- “Ip-Direct”
Avaya S8300B 8-25-58220 connection
Server Site
Avaya S8300B 40208 G.729
Server Site to x58220 x208 -- or -- “Ip-Direct”
DO i40 Site 8-40-208 connection
Avaya S8300B Digital 20203 G.729
Server Site to Telephone x203 -- or -- IP Telephone to
DO i120 Site x58203 8-20-203 Avaya G250
Media Gateway
Avaya S8300B 58430 G.729
Server Site to x58220 x58430 -- or -- “Ip-Direct”
Avaya S8300C 8-45-58430 connection
Server Site
Table 2 – Example Calls Involving the Avaya S8300B Server Introduced in Figure 2
The configuration of the site served by the Avaya Distributed Office i40 is also documented in
prior Application Notes [2]. Another example of inter-site dialing using a SIP private network,
where three digit prefix codes were used, is documented in [3]. Reference [3] also summarizes
SIP message flows.
For more information on alternate routing or LAR, please consult Avaya Communication
Manager documentation, such as Reference [7]. Calls that trigger SIP LAR in the sample
network are illustrated in the verifications in Section 8.5.
In general, the SIP private networking enhancements in Avaya Communication Manager Release
5 enable Avaya Communication Manager systems to be networked with SIP trunks directly,
Access the Avaya Distributed Office Central Manager by typing in the URL of the server hosting
the Avaya Integrated Management components into a web browser. The following screen is
presented.
Click the Add Row button. In the new row, enter a descriptive Name. Enter the IP Address
(e.g., 2.2.125.88) of the Avaya S8300B Server in the new site. Enter a unique Prefix (e.g., 25)
that other sites can use to reach the newly added site. Enter the Extension Length at this site. In
the sample network, the Avaya Communication Manager sites use a five-digit uniform dial plan.
Click Jobs to view the job status, as shown below. Observe the Active job (bottom of list).
Access the SES Edge server by entering http://<SES Edge server IP Address>/admin in a web
browser. Click Launch SES Administration Interface.
Optionally, to confirm the configuration, select Trusted Hosts Æ List. The trust relationship
can be observed in the following screen.
From the List Host Address Map screen below, click Add Map in New Group.
The following screen shows the Host Address Map Entry handling UDP dialed calls of the form
582XX. In the Pattern field, the following is entered: “^sip:582[0-9][0-9]@distributed.com”.
The following screen appears summarizing the new host address map configuration.
4.2.2. New Media Server Address Map for Co-Resident SES Home
A media server address map is added for the Co-Resident SES Home so that calls of the form
584XX are routed to the Co-resident Avaya Communication Manager on the Avaya S8300C
Server. This configuration mirrors the configuration from Reference [1] for seven digit calls
beginning with the branch prefix 45. Again, the following configuration would not be necessary
if Avaya Communication Manager on the newly introduced Avaya S8300B Server were
configured to insert branch prefix 45 on the route pattern for UDP-dialed calls to 584XX.
The following screen reflects the configuration from Reference [1]. This configuration mapped
calls beginning with the branch prefix 45 to the Co-resident Avaya Communication Manager
contact shown in the screen. Since this same “Contact” applies to the new UDP dialing pattern
(i.e., no branch prefix), click Add Another Map to the existing group.
The resultant screen below illustrates the Media Server Address Map configuration. Calls that
arrive with the branch prefix 45, as will be the case for all calls from the Avaya Distributed
Office sites, will be handed off to Avaya Communication Manager. Calls that arrive with the
UDP format 584XX will be handed off using this same contact.
Optionally, check the configuration by clicking Address Map Priorities. The following screen
summarizes the configuration.
The following entry in the AAR Analysis table will send AAR calls of the form 582XX to route
pattern 20. The AAR Analysis configuration, taken together with the UDP configuration in the
prior section, allows a user to dial 582XX, 8-582XX, or 8-25-582XX to reach users at the newly
added Avaya S8300B Server site.
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
The command “add trunk-group” can be used to add a trunk group associated with the SIP
signaling group. In the screen capture that follows, the Group Type is set to “sip”, the
Signaling Group is set to 2, the Service Type is set to “tie”, and an appropriate TAC and
Number of Members have been specified. Each connection arriving from another site via the
SES Edge will use one SIP trunk member. For example, referring to the extensions shown in
Figure 2, a call between x58430 and x58203 will use one member of trunk group 2. As another
example, a call between x203 at the Avaya Distributed Office i120 branch and x58220 will also
use one member of trunk group 2.
Non-IP devices (analog, digital) derive a network region from the region of the cabinet or
gateway to which the device is connected. For example, digital station 58203 shown in Figure 2
is connected to the Avaya G250 Media Gateway, and is in network region 1 because the Avaya
G250 Media Gateway is in network region 1. When a non-IP device such as a digital phone
makes a call that is routed over the SIP Trunk Group, a media processing resource (e.g., the
integrated VoIP processing of the Avaya G250 Media Gateway) is required.
The following screen illustrates Page 3 for network region 1. The connectivity between network
region 2 (calls using the SIP signaling group) and network region 1 (H.323 IP Telephones,
The assignment of codec set 2 between network region 1 and 2 automatically creates a
symmetrical configuration for region 2 (to region 1). Other default parameters are retained for
network region 2, including the settings allowing direct IP media for inter-region connections.
Media Encryption
1: aes
2: none
3:
IP Codec Set
Codec Set: 2
Media Encryption
1: none
2:
3:
The following screen shows example station configuration for H.323 telephone x58220.
change station 58220 Page 1 of 6
STATION
Extension: 58220 Lock Messages? n BCC: 0
Type: 9620 Security Code: 1234 TN: 1
Port: S00005 Coverage Path 1: 99 COR: 1
Name: John Zack Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 19 Personalized Ringing Pattern: 1
Message Lamp Ext: 58220
Speakerphone: 2-way Mute Button Enabled? y
Display Language: english
Survivable GK Node Name:
Survivable COR: internal Media Complex Ext:
Survivable Trunk Dest? y IP SoftPhone? n
On Page 2 for this IP Telephone station (not shown), the default configuration to allow Direct
IP-IP Audio Connections is retained, by leaving the default value “y”.
The following screens show feature button programming that will be used in conjunction with
the verification of the SIP private networking enhancements in Section 8.6.
Such a configuration is only practical if the Avaya Distributed Office prefix code plus extension
are configured as an extension within the Avaya Communication Manager dial plan. In the
sample configuration, five digit numbers beginning with 2, 4, and 5 are defined as extensions in
the Avaya Communication Manager dial plan. Since extensions of the form 20XXX and
40XXX are not local extensions, the UDP table is consulted.
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest next
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
Route pattern 2 contains SIP trunk group 2 as the first choice. If the call uses the SIP trunk
group, no digit manipulation is necessary, as a result of the address map configuration on the
Avaya SES Edge shown in Section 4. The SES Edge will route 584XX to the site with the
Avaya S8300C Server. The LAR field is set to “next” for the first preference. If Avaya
Communication Manager chooses the SIP Trunk Group (i.e., it is in-service with an idle
member), but fails to get a timely response to the outbound SIP INVITE, or gets specific SIP
error responses from the far-end (listed in Section 1.1), the “next” choice in the pattern can be
used to attempt to complete the call. In the sample configuration, the alternate route is a simple
analog trunk group. In practice, other types of trunks may be used.
In the example, the Facility Restriction Level (FRL) of the alternate route is set to “7”, a high
FRL. Look-ahead routing from the private SIP network to alternate PSTN routes could be
permitted for “privileged” users, but not permitted for less privileged users, to avoid consuming
public trunks for inter-site traffic. From the screen below, it can also be observed that digit
manipulation is performed to transform the UDP dialed private number to a number that will be
The following screen illustrates route pattern 45, which is used when a user dials 8-45-584XX
via AAR. Route pattern 45 also contains SIP trunk group 2 as the first choice. If the call uses
the SIP trunk group, no digit manipulation is necessary. The SES Edge will route 45-584XX to
the site with the Avaya S8300C Server. As in route pattern 2, the LAR field is set to “next” for
the first preference, and trunk group 3 is the next preference, with a high FRL. Similar to route
pattern 2, the digits sent out the analog trunk are manipulated so that the far-end will receive
997-325-84XX, but in this case, the two leading digits (i.e., the branch prefix 45) are deleted
before digit insertion.
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest next
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest none
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
The following screen illustrates route pattern 40, which is used when a user dials 40XXX or 8-
40-XXX.
BCC VALUE TSC CA-TSC ITC BCIE Service/Feature PARM No. Numbering LAR
0 1 2 M 4 W Request Dgts Format
Subaddress
1: y y y y y n n rest next
2: y y y y y n n rest none
3: y y y y y n n rest none
4: y y y y y n n rest none
5: y y y y y n n rest none
6: y y y y y n n rest none
Success 0
Note that Avaya 9600-Series IP Telephones running the SIP protocol acquire dial plan
information from the Avaya call server as part of the log-in process. If changes are made to the
allowed dial-plan after the SIP telephone has logged in, it will be necessary for the SIP telephone
to re-acquire the dial-plan information. The Figure 1 network configured in Reference [1]
included SIP telephones at the two Avaya Distributed Office sites. In Section 3, using Avaya
Distributed Office Central Manager, a new site was added to the private network, allowing a new
branch prefix 25. Subsequently, the Avaya telephones running the SIP protocol were restarted to
allow the telephones to dial through AAR to the new site following re-registration.
8. Verification
The illustrated configuration has been verified. As explained in the corresponding section in
Reference [1], calls were made from each telephone shown in Figure 1 to every other telephone
to confirm the expected behavior. In addition, a call was placed from a telephone at each site to
the voice messaging application at another site. The remote telephone could log in and interact
with the voice messaging telephone user interface, indicating proper post-answer digit collection.
Similar types of verifications were used for the new site shown in the upper left quadrant of
Figure 2. This section includes basic site-to-site SIP connectivity verifications as well as
verifications illustrating the two new SIP-related Avaya Communication Manager Release 5
features, SIP Look-Ahead Routing (Section 8.5), and SIP private networking feature
transparency (Section 8.6).
list registered-ip-stations
REGISTERED IP STATIONS
Station Ext/ Set Product Prod Station Net Gatekeeper TCP
Orig Port Type ID Rel IP Address Rgn IP Address Skt
58220 9620 IP_Phone 1.5000 2.2.1.202 1 2.2.125.88 y
The following screen shows an example output of “list media-gateway”, showing that the Avaya
G250 Media Gateway is registered along with other pertinent information. The IP Address of
the gateway is 2.2.125.87. The IP Address of the S8300B Server is 2.2.125.88.
list media-gateway
MEDIA-GATEWAY REPORT
Num Name Serial No/ IP Address/ Type NetRgn Reg?
FW Ver/HW Vint Cntrl IP Addr RecRule
1 G250-DCP-with-ICC 06IS03701214 2 .2 .125.87 g250-dcp 1 y
27 .26 .0 /2 2 .2 .125.88 none
display media-gateway 1
MEDIA GATEWAY
Number: 1 Registered? y
Type: g250-dcp FW Version/HW Vintage: 27 .26 .0 /2
Name: G250-DCP-with-ICC MGP IP Address: 2 .2 .125.87
Serial No: 06IS03701214 Controller IP Address: 2 .2 .125.88
Encrypt Link? y MAC Address: 00:04:0d:6d:54:d1
Network Region: 1
Location: 1 Site Data:
Recovery Rule: none
The following screens show how the routing configuration established in Section 6 can be
inspected using the “list aar route-chosen” command. If a user dials via AAR, 8-20202, the call
will be routed to route pattern 20. Although not used in these Application Notes, another Avaya
Communication Manager Release 5 enhancement, the ability to configure “location-based
AAR”, is also evident in this screen.
Similarly, if a user dials 8-40208, the call will be routed to route pattern 40.
If a user dials 8-25-58220, the call will be routed in the same fashion as an extension-dialed call
to x58220. This is a result of the AAR digit conversion configuration illustrated in Section 6.
list uniform-dialplan
UNIFORM DIAL PLAN TABLE
Matching Pattern Len Del Insert Digits Net Conv Node Num
20 5 0 aar n
40 5 0 aar n
584 5 0 aar n
8.2.1. List Trace Output for UDP Dialed Calls to Avaya S8300C Site Using
SIP Trunk
The following screen shows an example trace for a call from digital station 58203 to UDP
number 58430. The call routes to SIP trunk group 2 using route pattern 2. Observe the final
media path is G.729 between the Avaya G250 Media Gateway serving the local digital phone
(2.2.125.87) and the remote H.323 telephone (2.2.35.209).
The following screen shows abridged “status trunk” information for this call.
The following screen shows abridged “status trunk” information for this call.
The following screen shows the abridged “status trunk” information for this call.
The following screen shows the abridged “status trunk” information for this call.
The following screen shows the abridged “status trunk” information for this call.
The following screen shows the abridged “status trunk” information for this call.
The following screen shows the abridged “status trunk” information for this call.
In the following “status station” output, observe that the connection uses G.711MU and AES
media encryption, as configured for local calls via the codec set 1 configuration in Section 6.
The following shows a “status station” output for this call. The call is “ip-direct” from the local
H.323 station to the far-end gateway.
To perform a call trace, select Maintenance & Monitoring Æ Telephony Æ Users. Check the
box next to a user and click Trace Checked as shown below for the user with extension 202.
After the Clear button was used to clear the trace output, the following screen shows a call trace
for an incoming call from the newly added site. In this case, extension 58220 from Figure 2
dials 8-20-202. The final media path uses G.729/G.729A directly between the two IP
telephones.
Calls to and from SIP telephones can also be traced. Using the procedure illustrated previously,
the following screen shows a trace of extension 203, when extension 203 dials 8-25-58203. The
The following illustrates the “list trace” output for an inter-site call dialed via AAR from digital
station 58203 to 8-40-202. Prior to the traced call, eight private network calls were made to and
from the Avaya Distributed Office i40 branch such that all of the Avaya Distributed Office
private network SIP trunks were in use at the time of the traced call. Avaya Distributed Office
immediately rejects the traced call (“408 Request Timeout” message can be observed in SES
Edge traces). When this SIP message is received by the originating Avaya Communication
Manager, LAR is triggered. Although the SIP message name and denial event observed in the
Prior to the traced call below, the COR for station 58203 was changed such that the FRL
associated with the station’s COR was less than the FRL configured on the route pattern for the
second route pattern preference. Just prior to the traced call, a network failure is induced that
results in the originating Avaya Communication Manager receiving no proper SIP response to
the SIP INVITE message. As can be observed from the trace, the attempt to “look-ahead”
resulting from the timeout (after 4 seconds) is denied by the FRL restriction, and the end-user
The priority call behaviors described above have been verified. For example, referring to Figure
2, if the user of x58203 presses the priority feature button, and then dials 58430, or 8-45-58430,
and the call is routed over the SIP trunks, then station 58430 will ring with priority ringing
(audible and display). If x58430 does not answer, the priority call will not divert via call
coverage. That is, this inter-system call has priority call properties. Priority calls can be made
among any of the users in the upper quadrants of the Figure 2 diagram (i.e., users served by
Avaya Communication Manager).
Note that the use of priority calling is subject to Class of Service permissions, which are granted
to the users in the sample configuration.
For example, referring to Figure 2, assume a coverage path is defined such that when station
58203 does not answer, the call covers to station 58220. Assume the user of x58430 dials Janey
Digital at 58203, or 8-25-58203, and the call is routed over the SIP trunks, and Janey does not
answer. While Janey’s telephone is ringing, the caller’s telephone at x58430 will display “Janey
Digital 58203”. When coverage occurs, the caller’s display updates to show that coverage has
occurred. In this case, the display updates to “Janey Digital cover”. Assume that John Zack at
x58220 answers the covered call. The caller’s display updates with the identity of the answering
party, in this case, “John Zack 58220”.
For example, referring to Figure 2, assume a coverage path is defined such that when station
58220 does not answer, the call covers using “remote coverage” to station 58430. The
configuration is summarized in the screens below. Station 58220 is assigned coverage path 9.
COVERAGE POINTS
Terminate to Coverage Pts. with Bridged Appearances? n
Point1: r1 Rng: Point2:
Point3: Point4:
Point5: Point6:
The coverage remote table entry corresponding with “r1” contains the UDP destination 58430,
which will route to the distant Avaya Communication Manager via a SIP trunk.
Assume “Janey Digital” at x58203 dials local user “John Zack” at x58220, and John does not
answer so that call coverage on “don’t answer” criteria is met. The call is routed over the SIP
trunk to remote station 58430. The display on x58430 will show “Janey Digital to John Zack
d”. That is, the call history at the remote Avaya Communication Manager is displayed to the
party receiving the call from the SIP trunk between systems. “Janey Digital” is the caller, and
“John Zack” is the originally called party whose coverage path has been followed. The “d” is
the redirection reason for “don’t answer”. Hang up the call.
Now, make station 58220 “busy” with calls. That is, in this scenario, there are no idle call
appearances available for an incoming call to station 58220. Repeat the call scenario above
where x58203 dials x58220, and the call covers over the SIP trunks to x58430. When the call
rings in to x58430, x58430 will display “Janey Digital to John Zack b”. In this example, the
“b” is the redirection reason for “busy”.
For example, referring to Figure 2, assume extensions 58220 and 58203 are members of a call
pickup group. Assume the user of x58430 dials John Zack at 58220, and the call is routed over
the SIP trunks. While x58220 is ringing, the caller’s display at x58430 will be “John Zack
58220”. Now assume that Janey Digital at x58203 presses a call pickup button on the telephone
to answer or “pickup” the call. When distant call pickup occurs, the caller’s display updates with
the connected party information. In this case, the caller’s display updates with “Janey Digital
58203”. This same display behavior occurs with directed call pickup. For example, if Janey
Digital is removed from the call pickup group, Janey may use the directed call pickup feature to
pickup the call ringing at John Zack’s telephone, assuming Janey has the proper COR privileges.
The call forwarding behaviors described above have been verified. For example, referring to
Figure 2, assume Janey Digital at x58203 has activated call forwarding “all calls” to John Zack
at x58220. Assume the user of x58430 dials Janey at x58203, and the call is routed over the SIP
trunks. While x58220 is ringing with the forwarded call, the caller’s display at x58430 will be
“Janey Digital forward”. That is, the caller is aware of the call forwarding occurring at the far-
end. When x58220 answers the forwarded call, the caller’s display updates with the identity of
the answering party at the far-end. In this case, “John Zack forward” is displayed to the caller.
Note that the call forwarding features are subject to Class of Service permissions, which are
granted to the users in the sample configuration.
As another example, assume station user 58220 deactivates call forwarding all, and activates call
forwarding on “don’t answer” to x58430. Assume Janey Digital at x58203 dials local user
58220, and the call is forwarded over the SIP trunk to remote station 58430. x58430 will display
“Janey Digital to John Zack f”, the same display as the case where call forwarding all is used.
As shown in the following screen, a system parameter governs the display behavior for bridging.
If the Identity When Bridging parameter is set to “station” as shown in bold below, then the
identity of the station that answered the call using the bridged appearance is displayed to the
caller. If this parameter is set to “principal”, then the identity of the called party (i.e., the
“principal”) is displayed to the caller, even if the call was answered using a bridged appearance
from another station.
CPN/ANI/ICLID PARAMETERS
CPN/ANI/ICLID Replacement for Restricted Calls: Restricted
CPN/ANI/ICLID Replacement for Unavailable Calls:
DISPLAY TEXT
Identity When Bridging: station
For example, referring to Figure 2, assume extension 58203 has a bridged appearance for station
58220 (as shown in Section 6). Assume the user of x58430 dials John Zack at x58220, and the
call is routed over the SIP trunks. While x58220 is ringing, the caller’s display at x58430 will
be “John Zack 58220”. Now assume that Janey Digital at x58203 presses a bridged
appearance button for x58220 to answer the call via the bridged appearance. When the call is
answered, the remote caller’s display will depend on the Identity When Bridging parameter. If
the parameter is set to “station”, the display on x58430, the calling party, will update with the
identity of the answering party. In this case, the caller’s display would update with “Janey
Digital 58203”. If the parameter is set to “principal”, the calling party would continue to see
the identity of the called party, “John Zack 58220”, even after the call was answered at the
bridged appearance on station 58203.
It is also possible to receive a display update when a call is transferred to another system across
SIP trunks. For example, assume Janey Digital at x58203 calls local extension 58220. The user
at 58220 completes a transfer of the call across the SIP trunk to “Jim Essential” at 58430. The
display at x58430 shows “Janey Digital 58203”. The display at x58203 shows “Jim Essential
58430”. This is an example of a transfer to a user at another system that results in proper
displays when the inter-system transferred call uses a SIP trunk.
8.6.9. Display Updates for Conference Calls that Degenerate to 2 party Calls
When a conference occurs (3 or more parties in call), the display on the conference participants
shows a conference indication. When conference participants disconnect from the call leaving a
two-party call, the two remaining parties on the call can receive a display update with the
identity of the other remaining party.
For example, assume Jim Essential at x58430 dials Janey Digital at x58203, and the call is routed
over the SIP trunk between systems. Now assume Janey Digital conferences in local user John
Zack at x58220. The displays on all parties indicate a conference. Now assume Janey Digital
hangs up. The display on x58430 updates to “John Zack 58220”, and the display on x58220
updates to “Jim Essential 58430”, reflecting the two-party connection.
8.6.10. Calling Number Block and Un-block / Privacy For SIP Trunks
In these Application Notes, the calling party identity is conveyed and presented to the call
recipient when the call is routed over the SIP trunks. Avaya Communication Manager also has a
set of capabilities that enable a caller to remain “private”. Before dialing the destination, a user
can press a feature button (“cpn-blk”) or dial a corresponding feature access code to “block”
calling party presentation, when presentation would otherwise occur. If the desired default
behavior for a user is “privacy”, the originating user’s station record can restrict calling party
presentation. Another feature button (“cpn-unblk) or corresponding feature access code can be
used to “unblock” calling party presentation, allowing presentation when privacy would
otherwise result. With SIP private networking, the caller’s desire for privacy can be conveyed
over the SIP trunks to the receiving Avaya Communication Manager system.
The following example call scenarios illustrate the behavior. As shown in previous sections, if
Janey Digital at x58203 dials 58430, the display at user 58430 will show “Janey Digital 58203”
Hang up this call. Now assume Janey Digital presses a “cpn-blk” feature button before dialing
With modest additional configuration, the receiving Avaya Communication Manager system can
display a specific text string for incoming calls marked for privacy. For example, as shown in
the following screen, the word “restricted” is configured to appear for incoming calls marked for
privacy on the Avaya Communication Manager running in the Avaya S8300C Server. This is a
system parameter that will apply to incoming calls from other types of trunks as well, such as
ISDN-PRI trunks, which have long supported this set of features.
CPN/ANI/ICLID PARAMETERS
CPN/ANI/ICLID Replacement for Restricted Calls: restricted
CPN/ANI/ICLID Replacement for Unavailable Calls: unavailable
A field on the trunk group form, which is new to Avaya Communication Manager Release 5 for
SIP trunks, enables the system parameter text to apply for incoming SIP calls that are marked for
privacy. This field is shown in bold in the following screen.
With this programming in place, assume Janey Digital again presses a “cpn-blk” button before
dialing 58430. When the call rings in at x58430, the display will show “CALL FROM
restricted”.
With this change in place, if Janey Digital dials 58430, the display at x58430 will again be
“CALL FROM restricted”. That is, Janey’s trunk calls have privacy without pressing a feature
button or dialing a feature access code before dialing.
If the default behavior would yield “privacy”, but Janey wants her identity to be presented for a
specific call, Janey can use a “cpn-unblk” button or corresponding feature access code. If Janey
Digital presses a “cpn-unlbk” button and dials 58430, the display at x58430 would be “Janey
Digital 58203”.
9. Conclusion
As illustrated in these Application Notes, customers who prefer a local call server may have
some sites with Avaya Distributed Office and other sites with Avaya Communication Manager.
These sites can operate independently, but can also be networked via SIP private networking.
An Avaya SES Edge server can provide the Master Administration function for one or more SES
Home(s), including a Co-resident SES Home in an Avaya S8300C Server. This same SES Edge
can function as the “core router” directing SIP flows between sites for inter-site calling. With
Avaya Communication Manager Release 5, SIP private networking is enhanced. The inter-site
operation of features including priority calling, call coverage, call pickup, call forwarding, and
bridging are enhanced with respect to displays and other call behaviors. Look-Ahead Routing is
enhanced for SIP trunks such that calls delivered to the SIP private network can be automatically
re-routed in the event of network failures.
10. References
The Application Notes in Reference [1] detail the configuration of the network foundation shown
in Figure 1.
[1] Sample Configuration for SIP Private Networking among Avaya Distributed Office sites and
Avaya Communication Manager Release 5 with Co-resident SES Home, Issue 1.0 December
2007.
The Application Notes in Reference [2] detail the configuration of the Avaya Distributed Office
i40 site used in the sample configuration.
[2] Configuring Avaya Distributed Office for Key System Features including Outside Line
Groups, Voice Announce, and Busy Indication and Transfer, Issue 1.0, August 2007.
http://www.avaya.com/master-usa/en-us/resource/assets/applicationnotes/do-key.pdf
The Application Notes in Reference [3] cover another SIP private network with both Avaya
Communication Manager and Distributed Office sites, using three-digit prefix codes. Sample
SIP message flows are included.
[3] Sample Configuration for SIP Connectivity between Avaya Communication Manager and
Avaya Distributed Office Using Avaya SIP Enablement Services, Issue 1.0, August 2007.
References [4-6] are examples of relevant Avaya Distributed Office Release 1.1 product
documentation available at http://support.avaya.com
[4] Feature Description for Avaya Distributed Office, Document 03-602027, Issue 1.0, May
2007.
http://support.avaya.com/elmodocs2/distributedoffice/r1_1/03-602027.pdf
[5] Design and Implementation Guide for Avaya Distributed Office, Document 03-602023, Issue
1.0, May 2007.
http://support.avaya.com/elmodocs2/distributedoffice/r1_1/03-602023.pdf
[6] Maintenance and Troubleshooting Guide for Avaya Distributed Office, Document 03-
602029, Issue 1.0, May 2007.
http://support.avaya.com/elmodocs2/distributedoffice/r1_1/03-602029.pdf
[7] Administrators Guide for Avaya Communication Manager, Document 03-300509, Issue 4.0,
Release 5.0, Jan 2008.
http://support.avaya.com/elmodocs2/comm_mgr/r5.0/03-300509_4.pdf
A definition of different SES server types as well as an introduction to SIP and SIP Enablement
Services is given in Reference [8].
Please e-mail any questions or comments pertaining to these Application Notes along with the
full title name and filename, located in the lower right corner, directly to the Avaya Solution &
Interoperability Test Lab at interoplabnotes@list.avaya.com