SIP PVT LAR

Avaya Solution & Interoperability Test Lab
Sample Configuration for SIP Private Networking and SIP

Look-Ahead Routing Using Avaya Communication Manager
– Issue 1.0
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.
JRR; Reviewed: Solution & Interoperability Test Lab Application Notes 1 of 65

SPOC 2/25/2008 ©2008 Avaya Inc. All Rights Reserved. SIP-Pvt-LAR
1. Introduction and Scope
These Application Notes illustrate a sample configuration using a SIP private network for inter-
site calling among four sites. Figure 1 illustrates the network used to verify previously
published Application Notes [1] that describe the configuration and operation for three of the
four sites. One site uses an Avaya Distributed Office i40, and another uses an Avaya Distributed
Office i120. A third site is equipped with an Avaya G450 Media Gateway and Avaya S8300C
Server, running Avaya Communication Manager Release 5, the IA770 Messaging application,
and Co-resident SIP Enablement Services (SES). The Co-resident SES is configured as an SES
Home [8], communicating with an Avaya S8500C SES functioning as the SES Edge Master
Administrator as well as the “core router” for inter-site SIP private networking. These initial
three sites have at least one Avaya IP Telephone running H.323 firmware and at least one Avaya
IP Telephone running SIP firmware. Each telephone is registered with its local server, capable
of providing telephony and voice mail services. Beginning with Release 5, an Avaya S8300C
Server can be configured for co-residence of Avaya SES along with Avaya Communication
Manager and IA770 Messaging. The Co-resident SES can be configured as a combined
Home/Edge, or as a Home using a separate SES Release 5 Edge, with the latter illustrated in
these Application Notes.
Figure 1: Network Overview Repeated from Reference [1]

Although not the focus of these Application Notes, the Avaya G450 Media Gateway is also
introduced with Avaya Communication Manager Release 5, and used in the sample
configuration. The Co-resident SES with Avaya Communication Manager can apply to any
Avaya Media Gateway accepting an S8300C Server, including the Avaya G250, G350, G450,
and G700 Media Gateways.
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.
Figure 2: Network Overview

In the sample configuration, the two sites with Avaya Distributed Office use a default three digit
dial plan for extensions. The two sites served by Avaya Communication Manager use a five
digit dial plan for extensions. Users may dial other users at the same site by simply dialing the
local extension.
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.

Table 1, repeated from Reference [1], summarizes the dialing flexibility among the sites in the
Figure 1 network. Table 2 summarizes the new dialing options made possible by the
configuration presented in these Application Notes, relevant for the newly introduced site shown
in the upper left quadrant of Figure 2. In both tables, the “Final Result” column assumes the call
is completed using the SIP private network, and the called party answers the telephone. Detailed
verifications for a representative sample of calls are presented in Section 8.
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
Table 1 – Example Calls and Expected Results Repeated from Reference [1]

Table 2 summarizes the new dialing options relevant for the newly introduced Avaya S8300B
Server site shown in the upper left quadrant of Figure 2.
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
Avaya S8300B 8-25-58220 connection
Server Site
Avaya S8300B 40208 G.729
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
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.
1.1. Summary of Triggers for SIP Look-Ahead Routing

Look-Ahead Routing (LAR) has been a longstanding feature of Avaya Communication Manager,
initially developed for use with ISDN-PRI trunks, and subsequently extended to H.323 and SIP
trunks. If a call is delivered to an in-service/idle ISDN-PRI, H.323, or SIP trunk in a route
pattern, and network timeouts occur, or specific ISDN, H.323, or SIP messages are received
from the far-end, LAR is triggered. When LAR is triggered, Avaya Communication Manager
can be configured to either route the call to the next choice in the route pattern, or re-offer the

call to the same trunk one more time, before proceeding to next choices in the route pattern.
LAR can provide a form of “fast fail-over” to route around congested or failed network
conditions that have not yet rendered the trunks “out of service”. Strictly speaking, Look-Ahead
Routing for SIP Trunks is not new to Avaya Communication Manager Release 5, but Avaya
Communication Manager Release 5 enhances Look-Ahead Routing for SIP trunks, expanding
the conditions that would trigger LAR. LAR triggers for SIP trunks now include any of the
following:
• SIP INVITE message can not be transmitted

• INVITE is transmitted, and four seconds elapse before a proper SIP response is received
(i.e., no “SIP 100 Trying” response is received)
• INVITE is transmitted, and any of the following SIP messages are received:
o 403 Forbidden
o 406 Not Acceptable
o 408 Request Timeout
o 410 Gone
o 414 Request-URI Too Long
o 416 Unsupported URI Scheme
o 481 Call/Transaction Does Not Exist
o 482 Loop Detected
o 483 Too Many Hops
o 487 Request Terminated
o 5xx Server failure (all)
o 604 Does Not Exist Anywhere
o 606 Not Acceptable
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.
1.2. Feature Transparency among Networked Independent Systems

Avaya Communication Manager systems can be networked to enable certain features to operate
between independent systems similar to the way these same features operate within a single
Avaya Communication Manager system. The concept of offering “feature transparency”
between networked systems is certainly not new. For many years, independent systems have
been networked via ISDN-PRI trunks or H.323 trunks, and the QSIG protocol has enabled
certain features to work as if the independent networked systems were part of one larger system.
With Avaya Communication Manager Release 5, systems networked via SIP trunks can now
benefit from a set of SIP private networking enhancements. While a full description of the
features available between SIP-networked systems is beyond the scope of this document, Section
8.6 highlights expected call and display behaviors in the sample network for Priority Calling,
Call Coverage, Call Pickup, Call Forwarding, Bridged Call Appearance, and Call Transfer.
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,

without the need for an Avaya SES. In these Application Notes, the features are illustrated in the
context of a broader network that utilizes an SES Edge for routing inter-site traffic.
2. Equipment and Software Validated

Table 3 shows the equipment and version information used in the sample configuration shown in
Figure 2.
Network Component Version Information

Avaya Distributed Office AM110 in i40-DS1 1.1 SP1 (1.1.0_33.02-SP-1.0.0)
Avaya Distributed Office i40-DS1 27.12
Avaya Distributed Office AM110 in i120 1.1.1_41.03
Avaya Distributed Office i120 27.12
Avaya S8300C Server (Avaya G450 Media Gateway) Release 5.0 load
Avaya Communication Manager 825.4
Co-Resident SES 825.30
IA770 N5.0-12.0
Avaya G450 Media Gateway 27.26.0
Avaya S8300B Server (Avaya G250 Media Gateway) Avaya Communication Manager
Release 5.0 load 825.4
Avaya G250 Media Gateway 27.26.0
Avaya S8500C Server (SES Edge, SIP core router) SES R5 825.31
Avaya Distributed Office Central Manager 4.0.124.3098
Avaya 9600-Series IP Telephones (SIP) 1.0.13.1 (5)
Avaya 9600-Series IP Telephones (H.323) 1.5
Table 3 – Equipment and Version Information

3. SIP Private Networking Configuration Using Avaya
Distributed Office Central Manager
The corresponding section in Reference [1] illustrates the Avaya Distributed Office Central
Manager configuration for the SIP private networking illustrated in Figure 1. This section will
illustrate the additions to the Avaya Distributed Office Central Manager configuration related to
the introduction of the fourth site with the Avaya S8300B Server shown in the upper left
quadrant in Figure 2.
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 Distributed Office Central Manager link under the heading Provisioning and
configuration. Log into the application (login screen not shown). Click Enterprise Æ Private
Networking. Click the SIP Servers tab. The following screen is presented, reflecting prior
configuration illustrated in Reference [1].
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.

Press the Save Changes button. The following screen is presented for job scheduling. Enter
descriptive Job Notes, and click Run now, followed by the OK button.
Click Jobs to view the job status, as shown below. Observe the Active job (bottom of list).

The Refresh button can be used to check for job completion. Optionally, to verify the intended
effect, the Avaya SES Edge web pages may be used to confirm the new branch prefix
configuration, as shown in the verifications in Section 8.1.
4. Avaya S8500C SES Edge Server Configuration

The corresponding section in Reference [1] illustrates the Avaya S8500C SES Edge Server
configuration for the network illustrated in Figure 1. This section will illustrate the additions to
the configuration related to the introduction of the new site with the Avaya S8300B Server
shown in the upper left quadrant in Figure 2.
Access the SES Edge server by entering http://<SES Edge server IP Address>/admin in a web
browser. Click Launch SES Administration Interface.

4.1. Trusted Host Configuration
The IP Address of the Avaya S8300B Server will be added as a trusted host to the Avaya SES
Edge Server. From the SES Administration web interface, select Trusted Hosts Æ Add. As
shown in the following screen, select the IP Address of the SES Edge (2.2.166.13) from the Host
drop-down menu. In the IP Address field, enter the IP Address (2.2.125.88) of the Avaya
S8300B Server. In the Comment field, enter appropriate descriptive text. Click the Add button.
Optionally, to confirm the configuration, select Trusted Hosts Æ List. The trust relationship
can be observed in the following screen.
4.2. Address Map Configuration

With the network of Figure 1 in place, the SES Edge already had the configuration that would
allow the newly introduced Avaya S8300B Server to reach the other three sites. That is, if

Avaya Communication Manager on the Avaya S8300B Server were configured to ensure the
appropriate branch prefix codes were sent to the SES Edge for UDP as well as AAR dialing
sequences, the configuration from Reference [1] would enable outbound calls to terminate
properly at any of the other three sites. Similarly, if Avaya Communication Manager on the
Avaya S8300C Server were configured to send the prefix code of the “new” site with the Avaya
S8300B Server to the SES Edge for UDP and AAR dialed calls, then the configuration in Section
3 and Section 4.1 would be sufficient for inter-site calling.
In the interests of illustrating alternative approaches, in Section 6, Avaya Communication

Manager on the Avaya S8300B is configured such that UDP-dialed calls (584XX) to the site
with the Avaya S8300C Server will not insert the branch prefix code (i.e., 45) on the selected
route pattern. As a result, the SES Edge must recognize a new dial sequence of the form 584XX.
Similarly, in Section 5, Avaya Communication Manager on the Avaya S8300C Server will not
insert branch prefix code 25 on the selected route pattern for UDP dialed calls to 582XX. The
SES Edge will be configured with address maps to direct 584XX and 582XX calls to the
appropriate site.
4.2.1. New Host Address Map for SES Edge

From the SES Administration web interface, select Hosts Æ List. From the resulting list, select
Map for the row with the SES Edge (i.e., Host IP address 2.2.166.13).
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”.

In the Name field, enter appropriate descriptive text. In this case, since the pattern 582XX refers
to calls that will be sent to the site with the Avaya S8300B Server in Avaya G250 Media
Gateway, the text “To-S8300B-G250” is entered. The default Replace URI check is retained.
Click Add.
In the confirmation screen, click Continue.
In the resulting screen shown below, click Add Another Contact.

In the Contact field, enter “sip:$(user)@2.2.125.88:5061;transport=tls”. Click Add.
The following confirmation screen appears. Click Continue.
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.

From the SES Administration web interface, select Media Servers Æ List. From the resulting
list, select Map for the row with the Co-Resident SES Home (i.e., 1.1.45.88).
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.
In the Pattern field, the following is entered: “^sip:584[0-9][0-9]@distributed.com”. In the

Name field, enter appropriate descriptive text. In this case, since the pattern 584XX refers to
UDP-dialed calls that arrive from the site with the Avaya S8300B Server, the text “From-
S8300B-UDP” is entered. The default Replace URI check is retained. Click Add.

In the confirmation screen that follows, click Continue.
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.

5. Changed Avaya Communication Manager Configuration
for Avaya S8300C Server
The corresponding section in Reference [1] illustrates the configuration for the network shown in
Figure 1 and the dialing sequences shown in Table 1. This section will only include the
additions to routing to reach the new site shown in the upper left quadrant in Figure 2, enabling
the outbound dialing sequences from the Avaya S8300C Server shown in Table 2.
5.1. UDP Entry To Reach Users at Avaya S8300B Site

The users at the Avaya S8300B Server site have extensions of the form 582XX. The following
entry in the UDP table will send calls dialed as 582XX to AAR.
change uniform-dialplan 582 Page 1 of 2

UNIFORM DIAL PLAN TABLE
Percent Full: 0
Matching Insert Node
Pattern Len Del Digits Net Conv Num
582 5 0 aar n
5.2. AAR Entries To Reach Users at Avaya S8300B Site

In the following screen, the bold entry in the AAR Analysis table will send AAR calls of the
form 25-XXXXX to route pattern 20. The rows in the table that are not in bold reflect the prior
configuration from Reference [1]. Observe that the “branch prefix” codes 20 and 40 for the
Avaya Distributed Office i120 and i40 sites are also directed to route pattern 20. Calls to the
new Avaya Communication Manager site can be handled the same way as calls to the Avaya
Distributed Office sites from a routing point of view.
change aar analysis 2 Page 1 of 2

AAR DIGIT ANALYSIS TABLE
Location: all Percent Full: 2
Dialed Total Route Call Node ANI

String Min Max Pattern Type Num Reqd
20 5 5 20 aar n
25 7 7 20 aar n
40 5 5 20 aar n
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.


582 5 5 20 aar n

5.3. Route Pattern Used For Calls to the Avaya S8300B Site
Although materially unchanged from Reference [1], the following screen illustrates the route
pattern used for UDP and AAR dialed calls destined for the newly added Avaya S8300B Server
site. The various types of calls are all routed to trunk group 66, which is the SIP trunk group to
the Co-Resident SES Home Server in the Avaya S8300C Server. The Co-Resident SES Home
will pass calls to the Avaya S8500C SES Edge Server. The SES Edge will route the 582XX, 8-
582XX and 8-25-582XX dialed calls to the newly added Avaya S8300B Server site as a result of
the configuration shown in Section 3 and Section 4. Note that if Look-Ahead Routing were to
be used for outbound calls from the Avaya S8300C Server to other sites, it would be necessary to
route calls to different route patterns. Each route pattern would enable unique digit manipulation
for the alternate route to the site. An example configuration using unique route patterns and
Look-Ahead Routing is provided in Section 6 for outbound calls from the Avaya S8300B Server.
change route-pattern 20 Page 1 of 3

Pattern Number: 20 Pattern Name: Private-SIP
SCCAN? n Secure SIP? n
Grp FRL NPA Pfx Hop Toll No. Inserted DCS/ IXC
No Mrk Lmt List Del Digits QSIG
Dgts Intw
1: 66 0 n user
2: n user
3: n user
4: n user
5: n user
6: n user
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

6. Avaya Communication Manager Configuration for Avaya
S8300B Server in Newly Added Site
This section illustrates relevant configuration for Avaya Communication Manager running on the
Avaya S8300B Server within the Avaya G250 Media Gateway shown in Figure 2.
6.1. Node Names

Node names are logical mappings of names to IP addresses that are used in other configuration
screens, such as the near-end and far-end of a SIP signaling group. The node name “procr”
refers to the Avaya S8300B Server Ethernet. The node name “SES-Edge” refers to the IP
Address of the Avaya SES Edge.
change node-names ip Page 1 of 2

IP NODE NAMES
Name IP Address
SES-Edge 2.2.166.13
procr 2.2.125.88
6.2. SIP Signaling Group and Trunk Group

Configure the SIP signaling group using the “add signaling group” command. In the sample
configuration, signaling group 2 is created. The parameters relevant to these Application Notes
are shown in bold below. The Group Type is set to “sip”. The Transport Method is set to
“tls”. The Near-end Node Name is set to “procr”, the Avaya S8300B Server Ethernet. The
Far-end Node Name is set to “SES-Edge”, the node name assigned to the IP address of the SES
Edge. The Near-end Listen Port is set to 5061. The Far-end Listen Port is also 5061. In the
sample configuration, the Far-end Network Region is set to 2, a network region that is different
from the network region of the near-end devices. By optionally assigning a unique network
region to the far-end of the SIP signaling group, different codec set options may be used for calls
that use the SIP signaling group and trunk group. The Far-end Domain is set to a domain
consistent with the overall configuration. In this case, the domain is “distributed.com”. The
Direct IP-IP Audio Connections field is left at the default value of “y” to allow direct media
paths for calls using this signaling group.
add signaling-group 2 Page 1 of 1

SIGNALING GROUP
Group Number: 2 Group Type: sip

Transport Method: tls
IP Video? n
Near-end Node Name: procr Far-end Node Name: SES-Edge

Near-end Listen Port: 5061 Far-end Listen Port: 5061
Far-end Network Region: 2
Far-end Domain: distributed.com
Bypass If IP Threshold Exceeded? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? y
IP Audio Hairpinning? n
Enable Layer 3 Test? y
Session Establishment Timer(min): 3

In the sample configuration, the Avaya S8300B Server is directly SIP networked with the Avaya
SES Edge. Typically, when Avaya Communication Manager is connected to an Avaya SES
Server, it is networked to an SES Home, not directly to an SES Edge. However, since there will
be no Avaya IP Telephones running the SIP protocol at the Avaya S8300B Server site, and since
the network is assumed to already contain an SES Edge (from Reference [1] and Figure 1) the
newly introduced Avaya S8300B Server is networked directly to the SES Edge, which facilitates
inter-site calling to and from the broader network.
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.
add trunk-group 2 Page 1 of 21

TRUNK GROUP
Group Number: 2 Group Type: sip CDR Reports: y
Group Name: To-SES-Edge COR: 1 TN: 1 TAC: 102
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: tie Auth Code? n
Signaling Group: 2
Number of Members: 8
6.3. Association of Endpoints with Network Regions

IP telephones can be assigned a network region based on an IP address mapping. The following
screen illustrates a subset of the IP network map configuration used to verify these Application
Notes. In the sample configuration, the Avaya H.323 IP Telephones registered to the Avaya
S8300B Server are assigned to default network region 1. The IP Network Map configuration
shown below is not strictly necessary. Avaya IP Telephones at the newly added site are assigned
IP Addresses in the range shown below.
change ip-network-map Page 1 of 32
IP ADDRESS MAPPING
Emergency
Subnet Location
From IP Address (To IP Address or Mask) Region VLAN Extension
2 .2 .1 .200 2 .2 .1 .210 1 n
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 shows the configuration of the Avaya G250 Media Gateway. Note that the
gateway is configured in network region 1 (the default).
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
Slot Module Type Name

V1: S8300 ICC MM
V2:
V3: 4T+2L-Integ-Analog ANA IMM
V4: Integ-DCP DCP IMM
Max Survivable IP Ext: 8
V9: gateway-announcements ANN VMM
6.4. Network Region Configuration

The following screens illustrate the configuration for network region 1. As shown in the
previous section, all H.323 and digital endpoints served by the Avaya S8300B Server are in
network region 1. From Page 1, observe that the Codec Set to be used for connections within
region 1 is set to “1”. The Authoritative Domain matches the “distributed.com” domain also
configured for the SIP signaling group. The Intra-region IP-IP Direct Audio and Inter-region
IP-IP Direct Audio fields are left at the default “yes” values to allow direct media connections.
change ip-network-region 1 Page 1 of 19
IP NETWORK REGION
Region: 1
Location: 1 Authoritative Domain: distributed.com
Name: Home court
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 1 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 3329
DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y
Call Control PHB Value: 46 RTCP MONITOR SERVER PARAMETERS
Audio PHB Value: 46 Use Default Server Parameters? y
Video PHB Value: 26
802.1P/Q PARAMETERS
Call Control 802.1p Priority: 6
Audio 802.1p Priority: 6
Video 802.1p Priority: 5 AUDIO RESOURCE RESERVATION PARAMETERS
H.323 IP ENDPOINTS RSVP Enabled? n
H.323 Link Bounce Recovery? y
Idle Traffic Interval (sec): 20
Keep-Alive Interval (sec): 5
Keep-Alive Count: 5
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,

digital telephones) is specified via the Inter Network Region Connection Management table
beginning on Page 3. These connections will use codec set 2 as shown below.
change ip-network-region 1 Page 3 of 19
Inter Network Region Connection Management
src dst codec direct WAN-BW-limits Video Dyn

rgn rgn set WAN Units Total Norm Prio Shr Intervening-regions CAC IGAR
1 1 1
1 2 2 y NoLimit n
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.
6.5. Codec Set Configuration

The network region configuration allows configuration of a codec set to be used for intra-region
connections, and the option of a different codec set to be used for inter-region connections for
any given network region pair. Codec set 1, used for intra-region connections, is configured as
shown in the following screen. Intra-region calls will prefer G.722 or G.711MU over G.729,
which will be used for inter-region connections. To illustrate other possible differences in the
codec sets, “aes” is listed as an option under the Media Encryption heading in codec set 1.
Connections among local H.323 telephones, or among H.323 telephones and digital telephones,
will have AES media encryption applied. An example local connection is shown in Section
8.2.9.
change ip-codec-set 1 Page 1 of 2

IP Codec Set
Codec Set: 1
Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)
1: G.722-64K 2 20
2: G.711MU n 2 20
3: G.729 n 2 20
4:
5:
6:
7:
Media Encryption
1: aes
2: none
3:

In these Application Notes, the far-end of the SIP signaling group is network region 2. Calls
between network regions, including network region 1 and network region 2, are configured to
use codec set 2, shown in the following screen. All inter-site private network SIP calls will use
G.729.
change ip-codec-set 2 Page 1 of 2
IP Codec Set
Codec Set: 2
Audio Silence Frames Packet

Codec Suppression Per Pkt Size(ms)
1: G.729 n 2 20
2:
3:
4:
5:
6:
7:
Media Encryption
1: none
2:
3:
6.6. Stations for Telephone Extensions

This section illustrates aspects of the configuration for the telephone extensions in the upper left
quadrant in Figure 2. The station with extension 58203 is a digital telephone connected to port
1V401, with name “Janey Digital”. (The name is referenced in the display verifications in
Section 8.6).
change station 58203 Page 1 of 5
STATION
Extension: 58203 Lock Messages? n BCC: 0

Type: 2420 Security Code: 1234 TN: 1
Port: 001V401 Coverage Path 1: 1 COR: 1
Name: Janey Digital Coverage Path 2: COS: 1
Hunt-to Station:
STATION OPTIONS
Time of Day Lock Table:
Loss Group: 2 Personalized Ringing Pattern: 1
Data Option: none Message Lamp Ext: 58203
Speakerphone: 2-way Mute Button Enabled? y
Display Language: english Expansion Module? n
Survivable COR: internal Media Complex Ext:

Survivable Trunk Dest? y IP SoftPhone? y
IP Video Softphone? n
Customizable Labels? y

The following screen shows feature button programming that will be used in conjunction with
the verification of the SIP private networking enhancements in Section 8.6.
STATION
BUTTON ASSIGNMENTS
1: call-appr 5: priority
2: call-appr 6: call-pkup
3: call-appr 7: call-fwd Ext:
4: brdg-appr B:1 E:58220 8:
voice-mail Number: 58255
The following screen shows example station configuration for H.323 telephone x58220.
STATION
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.

BUTTON ASSIGNMENTS
1: call-appr 4: send-calls Ext:
2: call-appr 5: call-fwd Ext:
3: priority 6: call-pkup
voice-mail Number: 58255
6.7. Feature Access Codes

The feature access code for Automatic Alternate Routing (AAR) is configured to be “8”, the
same as the (default) AAR access code used with Avaya Distributed Office.
change feature-access-codes Page 1 of 6

FEATURE ACCESS CODE (FAC)
Auto Alternate Routing (AAR) Access Code: 8

6.8. AAR Configuration
When a user dials 8, the AAR access code, the digits following the AAR access code are
analyzed according to the configuration in this section. In the sample network, if a user dials 8-
20XXX, where X is any digit, the call will be directed to route pattern 20. If a user dials 8-
40XXX, where X is any digit, the call will be directed to route pattern 40. If a user dials 8-
45XXXXX, where X is any digit, the call will be directed to route pattern 45. If a user dials 8-
584XX (or 584XX through UDP), the call will be directed to route pattern 2. Although these
route patterns all use SIP trunk group 2 as the first choice, different route patterns have been
established to allow for different options for Look-Ahead Routing, and different digit
manipulation on each route pattern.

20 5 5 20 aar n
40 5 5 40 aar n
45 7 7 45 aar n
584 5 5 2 aar n
6.8.1. Optional AAR Digit Conversion and AAR Analysis Configuration

The optional configuration in this section enables a caller at the Avaya S8300B Server site to dial
the AAR access code (e.g., 8), the local prefix of the site (e.g., 25) and extension, and have the
call complete as if the extension were dialed directly. The bolded row in the AAR Digit
Conversion table shown below matches on a dial string such as 8-25-58220, deletes the leading 2
digits that represent the local prefix, and allows the call to be routed to the resultant extension.
change aar digit-conversion 2 Page 1 of 2

AAR DIGIT CONVERSION TABLE
Matching Pattern Min Max Del Replacement String Net Conv ANI Req
25 7 7 2 ext n n
6.9. UDP Configuration

The following screen shows a portion of the Uniform Dial Plan (UDP) configuration. If a user
dials the prefix code and extension of a user at an Avaya Distributed Office branch without
dialing the AAR access code first, the translations in the UDP table can allow the call to
complete. From the perspective of the Avaya Communication Manager user dialing the call, this
configuration can allow the Avaya Distributed Office sites to appear to be part of a broader UDP
network among other Avaya Communication Manager sites.
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.

The UDP table shown below sends the calls to AAR, which will handle the call as if the AAR
access code had been dialed.
change uniform-dialplan 0 Page 1 of 2

Percent Full: 0
Matching Insert Node
Pattern Len Del Digits Net Conv Num
584 5 0 aar n
20 5 0 aar n
40 5 0 aar n
6.10. Route Patterns With and Without Look-Ahead Routing

The following screen illustrates route pattern 2, which is used when a user dials 584XX via UDP.

Pattern Number: 2 Pattern Name: UDP-SIP-w-LAR
Dgts Intw
1: 2 0 n user
2: 3 7 99732 n user
3: n user
4: n user
5: n user
6: n user
Subaddress
1: y y y y y n n rest next
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

routable by the far-end of the analog trunk. A dialed number such as 58430 will be converted to
997-325-8430, which in the sample network can route to extension 58430 at the site with the
Avaya S8300C Server.
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.

Pattern Number: 45 Pattern Name: AAR-SIP-w-LAR
Dgts Intw
1: 2 0 n user
2: 3 7 2 99732 n user
3: n user
4: n user
5: n user
6: n user
Subaddress

The following screen illustrates route pattern 20, which is used when a user dials 20XXX or 8-
20-XXX. Route pattern 20 contains SIP trunk group 2 as the only choice. Look-ahead routing
could have been used as long as an alternate route exists to the destination Avaya Distributed
Office i120 site (e.g., through PSTN trunks). The LAR behavior occurring at the originating
Avaya Communication Manager would be no different than the LAR examples illustrated in
these Application Notes.

Pattern Number: 20 Pattern Name: To-i120-no-LAR
Dgts Intw
1: 2 0 n user
2: n user
3: n user
4: n user
5: n user
6: n user
Subaddress
The following screen illustrates route pattern 40, which is used when a user dials 40XXX or 8-
40-XXX.

Pattern Number: 40 Pattern Name: To-i40
Dgts Intw
1: 2 0 n user
2: 3 7 99555 n user
3: n user
4: n user
5: n user
6: n user
Subaddress

Route pattern 40 is similar to route pattern 2 and 45, where the first choice is a SIP trunk group
with “next” Look-Ahead Routing to analog trunk group 3, and digit manipulation allowing the
far-end of trunk group 3 to route the call. In these Application Notes, the purpose of configuring
LAR for the SIP route to Avaya Distributed Office was to illustrate a case where Avaya
Distributed Office rejects an incoming SIP private network call from the SES Edge, due to
configurable private call limits being reached. The SIP rejection by Avaya Distributed Office is
propagated back to the originating Avaya Communication Manager system, which can trigger
LAR processing to attempt to complete the call via a different trunk. See Section 8.5.
6.11. Incoming Call Handling Digit Manipulation for Trunk Group 2

When a user at any other site dials 8-25-582XX, the call will arrive to Avaya Communication
Manager from trunk group 2 and begin with the digits “25”. For example, when an Avaya
Distributed Office user dials 8-25-58203, the SES Edge will route the call to Avaya
Communication Manager on the Avaya S8300B Server. To enable proper Avaya
Communication Manager routing of such an inbound call, the incoming call handling treatment
for trunk group 2 will strip the digits “25” that represent the branch prefix. The resultant five
digit extension (e.g., 58203, 58220) will be routed by Avaya Communication Manager.
change inc-call-handling-trmt trunk-group 2 Page 1 of 3

INCOMING CALL HANDLING TREATMENT
Service/ Called Called Del Insert
Feature Len Number
tie 7 25 2
6.12. Public Unknown Numbering Configuration

The following screen illustrates a subset of the configuration that allows the calling and
connected number to be formatted for calls originated or answered by Avaya Communication
Manager users. The bold row below preserves the five digit extension, without modification, as
the calling or connected number for calls using SIP trunk group 2.
change public-unknown-numbering 0 Page 1 of 2

NUMBERING - PUBLIC/UNKNOWN FORMAT
Total
Ext Ext Trk CPN CPN
Len Code Grp(s) Prefix Len
Total Administered: 1
5 5 2 5 Maximum Entries: 240
6.13. Saving Configuration Changes

The command “save translation all” can be used to save configuration changes.
save translation all

SAVE TRANSLATION
Command Completion Status Error Code
Success 0

7. Avaya Distributed Office Local Configuration
The corresponding section in Reference [1] illustrates the optional Avaya Distributed Office
Local Manager configuration for the network shown in Figure 1. No changes are required using
Avaya Distributed Office Local Manager as a result of the introduction of the new site shown in
the upper left quadrant of Figure 2. The Avaya Distributed Office Central Manager
configuration illustrated in Section 3 populates each of the Avaya Distributed Office sites with
proper routing for calls of the form 8-25-XXXXX. Such calls are routed by the Avaya
Distributed Office branch sites to the SES Edge / Core Router, which in turn routes the calls to
the Avaya S8300B Server running Avaya Communication Manager at the new site in the upper
left quadrant of Figure 2.
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).

8.1. Verifications at the S8500C (SES Edge Server/Core Router)
To confirm that the core router function has been configured as expected via Avaya Distributed
Office Central Manager in Section 3, log in to the Avaya SES Edge Server as explained in
Section 4. Click Core Router Æ Prefix Map. The expected branch prefix codes, branch
addresses, and total lengths (prefix + extension) are shown.
8.2. Verifications from Avaya Communication Manager on the Avaya

S8300B Server
The following screen shows an abridged output of “list registered-ip-stations”, showing the
registered H.323 IP Telephone included in Figure 2.
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

The following screen shows an example output of “display media-gateway 1” illustrating other
aspects of the configuration. The digital station with extension 58203 in Figure 2 is connected
to port 1V401. The loop-start analog trunk is connected to port 1V301.
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
Slot Module Type Name

V1: S8300 ICC MM
V2:
V3: 4T+2L-Integ-Analog ANA IMM
V4: Integ-DCP DCP IMM
Max Survivable IP Ext: 8
V9: gateway-announcements ANN VMM
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.
list aar route-chosen 20202

AAR ROUTE CHOSEN REPORT
Location: all Partitioned Group Number: 1
Dialed Total Route Call Node
String Min Max Pattern Type Number Location
20 5 5 20 aar all
Similarly, if a user dials 8-40208, the call will be routed to route pattern 40.


40 5 5 40 aar all
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.

58220 5 5 ext all

If a user dials the same numbers illustrated in the previous screens, without dialing the AAR
access code, the call will route via UDP to the same route patterns. For example, if a user dials
40208, the call will be routed to route pattern 40, just as if the AAR access code had been dialed.
The UDP configuration (584xx) to reach the site with the Avaya S8300C Server can also be
observed.
list uniform-dialplan
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).
list trace station 58203 Page 1

LIST TRACE
time data
11:35:15 active station 58203 cid 0xa0
11:35:17 dial 58430 route:UDP|AAR
11:35:17 term trunk-group 2 cid 0xa0
11:35:17 route-pattern 2 preference 1 cid 0xa0
11:35:17 seize trunk-group 2 member 2 cid 0xa0
11:35:17 Calling Number & Name NO-CPNumber NO-CPName
11:35:17 Setup digits 58430
11:35:17 Calling Number & Name 58203 Janey Digital
11:35:17 Proceed trunk-group 2 member 2 cid 0xa0
11:35:18 Alert trunk-group 2 member 2 cid 0xa0
11:35:18 G729 ss:off ps:20 rn:2/1 1.1.45.87:2064 2.2.125.87:2050
11:35:18 xoip: fax:Relay modem:off tty:US 2.2.125.87:2050 uid:0x50006
11:35:20 active trunk-group 2 member 2 cid 0xa0
11:35:20 G729 ss:off ps:20 rn:2/1 2.2.35.209:2264 2.2.125.87:2050
The following screen shows abridged “status trunk” information for this call.
status trunk 2/2 Page 3 of 3

SRC PORT TO DEST PORT TALKPATH
src port: T00006
T00006:TX:2.2.35.209:2264/g729/20ms
001V022:RX:2.2.125.87:2050/g729/20ms:TX:ctxID:58
001V401:RX:ctxID:58

The following screen shows an example trace for a call from locally registered H.323 telephone
58220 to UDP number 58467, a SIP Telephone registered with the Co-resident SES Home in the
Avaya S8300C Server. The call routes to SIP trunk group 2 using route pattern 2. Observe the
final media path is G.729 between the two telephones (2.2.1.202, 2.2.66.107).

LIST TRACE
time data
11:41:35 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2050
11:41:35 xoip: fax:Relay modem:off tty:US 2.2.125.87:2050 uid:0x95d
11:41:36 Calling Number & Name 58220 John Zack
11:41:37 G729 ss:off ps:20 rn:2/1 1.1.45.87:2066 2.2.125.87:2052
11:41:44 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 1.1.45.87:2066
11:41:44 G729A ss:off ps:20 rn:2/1 1.1.45.87:2066 2.2.1.202:2756
11:41:44 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.66.107:5004
11:41:44 G729A ss:off ps:20 rn:2/1 2.2.66.107:5004 2.2.1.202:2756

8.2.2. List Trace Output for AAR Dialed Call to Avaya S8300C Site Using SIP
Trunk
The following screen shows an example trace for an inter-site call from H.323 station 58220 to
AAR number 8-45-58430. The call routes to SIP trunk group 2 using route pattern 45. Observe
the final media path uses G.729 between the locally registered H.323 telephone (2.2.1.202) and
the remote H.323 telephone (2.2.35.209).

LIST TRACE
time data
11:38:33 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2052
11:38:35 dial 845 route:AAR
11:38:37 dial 84558430 route:AAR
11:38:37 Setup digits 4558430
11:38:37 G729 ss:off ps:20 rn:2/1 1.1.45.87:2068 2.2.125.87:2054
11:38:45 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 1.1.45.87:2068
11:38:45 G729A ss:off ps:20 rn:2/1 1.1.45.87:2068 2.2.1.202:2756
11:38:45 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.35.209:2264
11:38:45 G729A ss:off ps:20 rn:2/1 2.2.35.209:2264 2.2.1.202:2756
The following screen shows abridged “status trunk” information for this call.

TRUNK STATUS
Trunk Group/Member: 0002/003 Service State: in-service/active

Port: T00007 Maintenance Busy? no
Connected Ports: S00005
Port Near-end IP Addr : Port Far-end IP Addr : Port

Signaling: 01A0017 2. 2.125. 88 : 5061 2. 2.166. 13 : 5061
G.729 Audio: 2. 2. 1.202 : 2756 2. 2. 35.209 : 2264
Audio Connection Type: ip-direct

8.2.3. List Trace Output for Incoming Call from Site with Avaya S8300C
The following screen shows an example output of “list trace tac 102” for an incoming call from a
phone at the site with the Avaya S8300C Server. In this case, extension 58430 dialed 58220 via
UDP. Although not presented, the same results would be obtained if extension 58430 dialed 8-
25-58220. Observe the final media path is G.729 directly between the calling telephone
(2.2.35.209) and the called telephone (2.2.1.202).
list trace tac 102 Page 1

LIST TRACE
time data
11:59:45 Calling party trunk-group 2 member 1 cid 0xa8
11:59:45 G729 ss:off ps:20 rn:2/1 1.1.45.87:2052 2.2.125.87:2054
11:59:45 dial 58220
11:59:45 ring station 58220 cid 0xa8
11:59:45 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2052
11:59:56 G729A ss:off ps:20 rn:2/1 1.1.45.87:2052 2.2.1.202:2756
11:59:56 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 1.1.45.87:2052
11:59:56 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.35.209:2264
11:59:56 G729A ss:off ps:20 rn:2/1 2.2.35.209:2264 2.2.1.202:2756
The following screen shows the abridged “status trunk” information for this call.

TRUNK STATUS
Signaling: 01A0017 2. 2.125. 88 : 5061 2. 2.166. 13 : 5061
G.729 Audio: 2. 2. 1.202 : 2756 2. 2. 35.209 : 2264

8.2.4. List Trace Output for Incoming Call from a Distributed Office Branch
The following screen shows an example output of “list trace tac 102” for an incoming call from a
phone at a Distributed Office site (x208 at the i40 site dialed 8-25-58220). Such a call uses one
SIP trunk member for the inbound call. Observe the final media path uses G.729 directly
between the calling telephone (2.2.40.104) and the called telephone (2.2.1.202).
list trace tac 102 Page 1

LIST TRACE
time data
15:52:58 Calling party trunk-group 2 member 1 cid 0xb1
15:52:58 active trunk-group 2 member 1 cid 0xb1
15:52:58 dial 58220
15:52:58 ring station 58220 cid 0xb1
15:52:58 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2050
15:52:58 G729 ss:off ps:20 rn:2/1 2.2.40.87:2596 2.2.125.87:2054
15:53:03 active station 58220 cid 0xb1
15:53:03 G729 ss:off ps:20 rn:2/1 2.2.40.104:2350 2.2.125.87:2054
15:53:03 G729A ss:off ps:20 rn:2/1 2.2.40.104:2350 2.2.1.202:2756
15:53:03 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.40.104:2350

TRUNK STATUS

Signaling: 01A0017 2. 2.125. 88 : 5061 2. 2.166. 13 : 5061
G.729 Audio: 2. 2. 1.202 : 2756 2. 2. 40.104 : 2350

8.2.5. List Trace Output for AAR dialed call to Distributed Office i40 Branch
The following screen shows an example output of “list trace station” for an outgoing call from
local digital telephone 58203. In this example, the caller dialed 8-40-208 to reach the user with
extension 208 at the Distributed Office site with branch prefix 40. The call routes to SIP trunk
group 2 using route pattern 40. Observe the final media path uses G.729 between the Avaya
G250 Media Gateway serving the local digital telephone (2.2.125.87) and the remote telephone
(2.2.40.104).

LIST TRACE
time data
11:44:27 G729 ss:off ps:20 rn:2/1 2.2.40.87:2570 2.2.125.87:2054
11:44:31 G729 ss:off ps:20 rn:2/1 2.2.40.104:2808 2.2.125.87:2054

TRUNK STATUS
Connected Ports: 001V401

Signaling: 01A0017 2. 2.125. 88 : 5061 2. 2.166. 13 : 5061
G.729 Audio: 1 2. 2.125. 87 : 2054 2. 2. 40.104 : 2808

8.2.6. List Trace Output for UDP dialed call to Distributed Office i40 Branch
The following screen shows an example output of “list trace station” for an outgoing UDP-dialed
call from local H.323 telephone 58220 to a user at the Avaya Distributed Office i40 site. In this
example, the caller dialed 40220 to reach the user with extension 220. This call is completed due
to the UDP configuration presented in Section 6. The call routes to SIP trunk group 2 using
route pattern 40. The final media path uses G.729 between the two telephones (2.2.1.202,
2.2.66.103).

LIST TRACE
time data
11:47:37 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2050
11:47:39 G729 ss:off ps:20 rn:2/1 2.2.40.87:2574 2.2.125.87:2052
11:47:45 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.40.87:2574
11:47:45 G729A ss:off ps:20 rn:2/1 2.2.40.87:2574 2.2.1.202:2756
11:47:46 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.66.103:5004
11:47:46 G729A ss:off ps:20 rn:2/1 2.2.66.103:5004 2.2.1.202:2756

8.2.7. List Trace Output for AAR dialed call to Distributed Office i120
Branch
The following screen shows an example output of “list trace station” for an outgoing call from
local digital telephone 58203. In this example, the caller dialed 8-20-203 to reach the user with
extension 203 at the Distributed Office i120 site with branch prefix 20. The call routes to SIP
trunk group 2 using route pattern 20. Observe the final media path uses G.729 between the
Avaya G250 Media Gateway serving the local digital telephone (2.2.125.87) and the remote
telephone (2.2.120.102).

LIST TRACE
time data
11:50:22 G729 ss:off ps:20 rn:2/1 2.2.120.87:2236 2.2.125.87:2050
11:50:30 G729 ss:off ps:20 rn:2/1 2.2.120.102:5004 2.2.125.87:2050

TRUNK STATUS
Connected Ports: 001V401

Signaling: 01A0017 2. 2.125. 88 : 5061 2. 2.166. 13 : 5061
G.729 Audio: 1 2. 2.125. 87 : 2050 2. 2.120.102 : 5004

8.2.8. List Trace Output for UDP dialed call to Distributed Office i120 Branch
The following screen shows an example output of “list trace station” for an outgoing UDP-dialed
call from local H.323 telephone 58220 to a user at the Avaya Distributed Office i120 site. In this
example, the caller dialed 20202 to reach the user with extension 202. This call is completed due
to the UDP configuration presented in Section 6. The call routes to SIP trunk group 2 using
route pattern 20. As with calls illustrated previously, the final media path uses G.729 directly
between the two telephones (2.2.1.202, 2.2.120.101).

LIST TRACE
time data
11:52:53 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2052
11:52:56 G729 ss:off ps:20 rn:2/1 2.2.120.87:2240 2.2.125.87:2054
11:53:00 G729 ss:off ps:20 rn:2/1 2.2.120.101:2814 2.2.125.87:2054
11:53:00 G729 ss:off ps:20 rn:1/2 2.2.1.202:2756 2.2.120.101:2814
11:53:00 G729A ss:off ps:20 rn:2/1 2.2.120.101:2814 2.2.1.202:2756

TRUNK STATUS

Signaling: 01A0017 2. 2.125. 88 : 5061 2. 2.166. 13 : 5061
G.729 Audio: 2. 2. 1.202 : 2756 2. 2.120.101 : 2814

8.2.9. List Trace Output for Calls Dialed Using AAR with Local 25 Prefix
The following screen shows an example output of “list trace station” for a call from a local
station to another local station dialed using AAR. Instead of dialing the local extension directly,
the user dialed the AAR access code followed by the prefix for the local Avaya Communication
Manager site, followed by the extension. Although the call was dialed via AAR, the call is
handled as a local call, as a result of the optional AAR Digit Conversion configuration shown in
Section 6. The final connection is identical to a call dialed by extension. The optional
configuration may be helpful to preserve familiar dialing patterns for mobile workers, who may
be accustomed to dialing using AAR and prefix codes from Avaya Distributed Office sites.

LIST TRACE
time data
14:52:12 active station 58203 cid 0x89
14:52:18 dial 82558220 route:AAR
14:52:18 ring station 58220 cid 0x89
14:52:18 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2134 2.2.125.87:2050
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.
status station 58203 Page 7 of 8

SRC PORT TO DEST PORT TALKPATH
src port: 001V401
001V401:TX:ctxID:6
001V022:RX:ctxID:6:TX:2.2.125.87:2050/g711u/20ms/aes
S00005:RX:2.2.1.202:2134/g711u/20ms/aes
dst port: S00005

8.3. Verification Steps From Avaya Communication Manager on the
S8300C Server
8.3.1. List Trace Output for UDP Dialed Call to Avaya S8300B Site
The following screen shows an example trace for a call from station 58430 to UDP number
58220. The call routes to SIP trunk group 66 using route pattern 20. Observe the final media
path uses G.729 between the two telephones (2.2.35.209, 2.2.1.202).

LIST TRACE
time data
11:42:58 active station 58430 cid 0x2cb
11:42:58 G711MU ss:off ps:20 rn:1/1 2.2.35.209:2264 1.1.45.87:2054
11:42:58 xoip: fax:Relay modem:off tty:US 1.1.45.87:2054 uid:0x95f
11:43:00 term trunk-group 66 cid 0x2cb
11:43:00 route-pattern 20 preference 1 cid 0x2cb
11:43:00 seize trunk-group 66 member 8 cid 0x2cb
11:43:00 Calling Number & Name 58430 Jim Essential
11:43:00 Proceed trunk-group 66 member 8 cid 0x2cb
11:43:00 Alert trunk-group 66 member 8 cid 0x2cb
11:43:00 G729 ss:off ps:20 rn:66/1 2.2.125.87:2050 1.1.45.87:2062
11:43:09 active trunk-group 66 member 8 cid 0x2cb
11:43:09 G729 ss:off ps:20 rn:66/1 2.2.1.202:2756 1.1.45.87:2062
11:43:09 G729 ss:off ps:20 rn:1/66 2.2.35.209:2264 2.2.1.202:2756
11:43:09 G729A ss:off ps:20 rn:66/1 2.2.1.202:2756 2.2.35.209:2264

8.3.2. List Trace Output for AAR Dialed Call to Avaya S8300B Site
The following screen shows an example trace for a call from station 58430 to AAR number 8-
25-58203. The call routes to SIP trunk group 66 using route pattern 20. Observe the final media
path is G.729 between the local telephones (2.2.35.209) and the remote Avaya G250 Media
Gateway (2.2.125.87) serving the called digital telephone. Note that if it is desired to use Look-
Ahead Routing, it may be necessary to route UDP-dialed and AAR-dialed calls to different route
patterns, so that unique digit manipulation can be performed for the alternate routes. (This is
illustrated for outbound calls from the Avaya S8300B Server in Section 8.5)

LIST TRACE
time data
15:34:16 active station 58430 cid 0x1a0
15:34:16 G711MU ss:off ps:20 rn:1/1 2.2.35.209:2264 1.1.45.87:2052
15:34:16 xoip: fax:Relay modem:off tty:US 1.1.45.87:2052 uid:0x95f
15:34:17 term trunk-group 66 cid 0x1a0
15:34:19 dial 82558203 route:AAR
15:34:19 route-pattern 20 preference 1 cid 0x1a0
15:34:19 seize trunk-group 66 member 7 cid 0x1a0
15:34:19 Setup digits 2558203
15:34:19 Calling Number & Name 58430 Jim Essential
15:34:19 Proceed trunk-group 66 member 7 cid 0x1a0
15:34:20 Alert trunk-group 66 member 7 cid 0x1a0
15:34:20 G729 ss:off ps:20 rn:66/1 2.2.125.87:2050 1.1.45.87:2050
15:34:26 active trunk-group 66 member 7 cid 0x1a0
15:34:26 G729 ss:off ps:20 rn:1/66 2.2.35.209:2264 2.2.125.87:2050
15:34:26 G729A ss:off ps:20 rn:66/1 2.2.125.87:2050 2.2.35.209:2264
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.
status station 58430 Page 5 of 8
AUDIO CHANNEL Port: S00005

G.729A Switch-End Audio Location:
IP Address Port Node Name Rgn
Other-End: 2. 2.125. 87 2050 66
Set-End: 2. 2. 35.209 2264 1

8.4. Verifications from Avaya Distributed Office Local Manager
The Avaya Distributed Office Local Manager interface can be accessed via a web browser using
the URL of the Avaya Distributed Office AM110 Application Module. In Avaya Distributed
Office R1.1.1, new capabilities for monitoring and tracing were added. In the sample
configuration, the AM110 in the Avaya Distributed Office i120 is running Release 1.1.1_41.03.
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.
In the resulting screen, click the Start button to begin a trace.

The following screen shows an example trace. In this case, the user of extension 202 dialed 8-
25-58220 to reach extension 58220 at the newly added Avaya Communication Manager site in
the upper left quadrant of Figure 2. It can be observed that the final media path uses
G.729/G.729A directly between the two IP telephones (2.2.120.101, 2.2.1.202).

After the Clear button was used to clear the trace output, the following screen shows a call trace
when the user of extension 202 dials 8-25-58203, a digital telephone at the same site as 58220.
The final media path uses G.729 between the remote Avaya G250 Media Gateway (2.2.125.87)
serving the digital telephone, and the IP Telephone (2.2.120.101) at the Avaya Distributed Office
site.
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

final media path uses G.729 between the remote Avaya G250 Media Gateway (2.2.125.87)
serving the digital telephone, and the SIP Telephone (2.2.120.102) at the Avaya Distributed
Office site.

8.5. Verifications from Avaya S8300B Server Associated with SIP
Look-Ahead Routing
In the following sub-sections, several different triggers for Look-Ahead Routing are illustrated.
8.5.1. Look-Ahead Routing Triggered by Lack of Timely Response to SIP

INVITE Messages
The following illustrates an annotated “list trace” output for an inter-site call from station 58220
to UDP extension 58430. The call is routed to route pattern 2. Just prior to the test, a transient
network failure is induced so that Avaya Communication Manager would not receive a timely
response to the outbound SIP INVITE message(s). As can be seen from the timestamps in the
annotated trace, after four seconds have elapsed without a proper response to the SIP INVITE,
Avaya Communication Manager “looks-ahead” to the next preference in the route-pattern. In the
sample configuration, the next preference is an analog trunk in the Avaya G250 Media Gateway.
The route-pattern includes digit manipulation that inserts the equivalent PSTN telephone number
for the user with extension 58430.

LIST TRACE
time data
16:36:49 G711MU ss:off ps:20 rn:1/1 2.2.1.202:2756 2.2.125.87:2050
16:36:51 term trunk-group 2 cid 0xb7
16:36:51 route-pattern 2 preference 1 cid 0xb7
16:36:51 seize trunk-group 2 member 4 cid 0xb7
16:36:51 Calling Number & Name 58220 John Zack /* SIP INVITE SENT */
/* NO proper response to SIP INVITE received. After 4 seconds, LAR is triggered */
16:36:55 denial event 1191: Network failure D1=0x95d D2=0x26
16:36:59 outpulse done 9973258430

The following illustrates the “list trace” output for an inter-site call dialed via AAR from digital
station 58203 to destination 8-45-58467. The call is routed to route pattern 45. Again, just prior
to the test, a transient network failure is induced so that Avaya Communication Manager would
not receive a timely response to the outbound SIP INVITE message(s). This trigger and trace are
similar to the previous, except that the call was dialed via AAR rather than UDP, and the AAR
call has been directed to a different route pattern.

LIST TRACE
time data
16:40:02 dial 84558467 route:AAR
16:40:02 Setup digits 4558467
16:40:06 denial event 1191: Network failure D1=0x95f D2=0x26
16:40:06 dial 84558467 route:AAR
16:40:06 dial 84558467 route:AAR
16:40:10 dial 84558467 route:AAR
16:40:10 outpulse done 9973258467
8.5.2. Look-Ahead Routing Triggered by Call Rejection at a Terminating

Distributed Office Branch
In Reference [1], the Avaya Distributed Office Central Manager was used to configure the
branches in the SIP private network. The number of simultaneous calls allowed to use the SIP
private network to an Avaya Distributed Office branch can be configured. In Section 3 of
Reference [1], the branch with the Avaya Distributed Office i40 was configured to allow 8 calls
to use the SIP private network. The following trace shows how Look-Ahead Routing occurring
at an originating Avaya Communication Manager site can re-route a call that is ultimately
rejected by the terminating Avaya Distributed Office site due to a configured maximum for
private network trunk calls. In the sample network, the alternate routing to the destination
telephone is not actually in place. This test and trace are included simply to illustrate the
triggering of LAR under the condition of “all SIP trunks busy to destination Avaya Distributed
Office”.
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

trace below imply a time lapse, the timestamps show that there is no perceivable delay before
LAR is triggered (i.e., the SIP INVITE and LAR trigger both have timestamp 16:02:18).

LIST TRACE
time data
16:02:12 term trunk-group 2 cid 0x53
16:02:18 route-pattern 40 preference 1 cid 0x53
16:02:18 seize trunk-group 2 member 8 cid 0x53
16:02:18 Proceed trunk-group 2 member 8 cid 0x53
16:02:18 denial event 1220: Recovery on timer expiry D1=0x95f D2=0x66
16:02:22 outpulse done 9955540202

8.5.3. Look-Ahead Routing Triggered by Call Rejection at the Terminating
Avaya Communication Manager Branch
station 58203 to 8-45-58420. Prior to the traced call, all members of the SIP trunk group at the
destination Avaya S8300C Server were in use handling other calls. Under these conditions, the
originating Avaya Communication Manager receives a “500 Server Internal Error” SIP message.
SIP Protocol traces (not shown) reveal that the terminating Avaya Communication Manager
responds with a “503 Service Unavailable” message and the Avaya SES Edge responds back to
the originating Avaya Communication Manager with a “500 Server Internal Error” SIP message.
When this SIP message is received by the originating Avaya Communication Manager, LAR is
triggered. Again, there is no perceivable delay before LAR is triggered (i.e., the SIP INVITE
and LAR trigger both have timestamp 15:11:36).

LIST TRACE
time data
15:11:36 dial 84558420 route:AAR
15:11:36 Setup digits 4558420
15:11:36 Proceed trunk-group 2 member 7 cid 0x51
15:11:36 denial event 1192: Temporary failure D1=0x95f D2=0x29
15:11:36 dial 84558420 route:AAR
15:11:36 dial 84558420 route:AAR
15:11:40 dial 84558420 route:AAR
15:11:40 outpulse done 9973258420
15:11:53 active trunk-group 3 member 1 cid 0x51

8.5.4. Look-Ahead Routing Triggered by Routing Loop / Too Many Hops
In the example below, the call was delivered to the SIP private network after the routing
configuration presented in these Application Notes was changed so that there was improper
configuration. The net result was that hop count thresholds were exceeded, so that the
originating Avaya Communication Manager system received a “483 Too Many Hops” SIP
message roughly 2 seconds after the initial SIP INVITE was sent. Since this message is also
among those that trigger LAR (see Section 1.1), the call “looks-ahead” to the next route pattern
preference. As explained along with the traces for other LAR triggers, the call is completed
over the analog trunk in the Avaya G250 Media Gateway.

LIST TRACE
time data
16:16:38 Proceed trunk-group 2 member 3 cid 0xb5
16:16:45 outpulse done 9973258430
8.5.5. Look-Ahead Routing As a Privilege, Restricting Use by FRL

The Facility Restriction Level (FRL) associated with a route pattern preference can be used to
restrict calls using the route pattern preference to privileged users. In the context of LAR, if the
SIP private network is the first choice route for inter-site calls, and the second choice route is a
public trunk, it may be desirable to limit access to the public trunk. Users with a Class of
Restriction (COR) that have an FRL below the FRL of the “public” route-pattern preference can
be denied access. Other users that have a COR with a sufficiently high FRL could dial the same
number and have the call complete. Although this is not a new concept, this section is included
to verify that expected behaviors also apply to calls using SIP LAR.
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

receives reorder tone. Other more privileged users, such as station 58220, could make a call at
this time, and the call would complete using LAR as shown previously in Section 8.5.1.

LIST TRACE
time data
16:44:19 active station 58203 cid 0xba
16:44:20 term trunk-group 2 cid 0xba
16:44:21 route-pattern 2 preference 1 cid 0xba
16:44:21 seize trunk-group 2 member 7 cid 0xba
16:44:25 denial event 1012: Destination Unavailable D1=0x95f D2=0xba
16:44:25 reorder trunk-group 2 cid 0xba
16:44:32 idle station 58203 cid 0xba
8.6. Verifications Associated with SIP Private Networking Feature

Transparency
This section describes the behavior of example calls that can benefit from the SIP private
networking enhancements introduced in Avaya Communication Manager Release 5. The display
behavior in this section is not meant to be prescriptive, and this section is not intended to be an
exhaustive list of the SIP private networking enhancements. The calls in this section assume the
call is routed using the SIP trunks, not using an alternate route.
8.6.1. Priority Calls between Avaya Communication Manager Sites

A caller can make a “priority call” by dialing a priority call feature access code, or by pressing a
priority feature button prior to dialing the called number. When a priority call rings in to the
called telephone, it has special properties, including priority ringing (audible and display),
overrides of restrictions that might otherwise prevent a lone remaining call appearance from
ringing, and overrides of certain diversion features such as call coverage that might otherwise
cause the call to leave the called telephone. The SIP private networking enhancements allow
priority calls to be made between users controlled by independent Avaya Communication
Manager Release 5 systems networked via SIP trunks.
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.

8.6.2. Display of Coverage Indication and Connected Party Identity to
Originator of Call, Distant Call Coverage
The call coverage feature allows a call to follow a “coverage path”. The SIP Private Networking
enhancements allow the caller to be made aware that call coverage has occurred, even when the
call coverage occurs at another Avaya Communication Manager Release 5 system networked
with the callers Avaya Communication Manager Release 5 system, via SIP trunks. A call that
follows a coverage path may also be answered by a user other than the called user. The SIP
Private Networking enhancements allow the caller to see the identity of the answering party for a
covered call.
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”.
8.6.3. Display of Redirection Reason At Coverage Point, Call Coverage

Between Systems Networked via SIP
This section focuses on the displays that appear on a station receiving a call that has covered
from another Avaya Communication Manager Release 5 system. Call coverage can be based on
different criteria, such as “busy” or “no answer”. The SIP private networking enhancements
allow the party receiving an incoming call to see the identity of the originally called party and
the associated redirection reason, even when the call coverage has occurred at another Avaya
Communication Manager Release 5 system networked via SIP trunks.
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.

STATION
Type: 9620 Security Code: **** TN: 1
Port: S00005 Coverage Path 1: 9 COR: 1
Name: John Zack Coverage Path 2: COS: 1
Hunt-to Station:

Coverage path 9 includes remote coverage entry “r1”, and various allowed coverage criteria.
change coverage path 9 Page 1 of 1

COVERAGE PATH
Coverage Path Number: 9
Hunt after Coverage? n
Next Path Number: Linkage
COVERAGE CRITERIA
Station/Group Status Inside Call Outside Call
Active? n n
Busy? y y
Don't Answer? y y Number of Rings: 2
All? n n
DND/SAC/Goto Cover? y y
Holiday Coverage? n n
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.
change coverage remote 1 Page 1 of 23

REMOTE CALL COVERAGE TABLE
ENTRIES FROM 1 TO 1000
01: 58430 16: 31:
02: 17: 32:
03: 18: 33:
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”.
8.6.4. Display of Connected Party Identity to Originator of Call, Distant Call

Pickup
The basic call pickup feature allows users that are members of a configured call pickup group to
“pickup” calls ringing at other telephones in the call pickup group. Directed call pickup allows
properly privileged users to pickup calls ringing at other telephones, without requiring any
configured group relationship. Call pickup is another example of a feature that would allow a

party to answer a call that is not the originally called party. The SIP private networking
enhancements allow the caller to see the identity of the answering party for a call that has been
answered using call pickup, when SIP networking is used between systems.
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.
8.6.5. Display of Forwarding Indication and Connected Party Identity to

Originator of Call, Distant Call Forwarding
Call forwarding allows a call to divert to a user-configurable alternate destination. The SIP
private networking enhancements allow the caller to be made aware that call forwarding has
occurred, even when the call forwarding occurs at another Avaya Communication Manager
Release 5 system networked with the callers Avaya Communication Manager Release 5 system,
via SIP trunks. Call forwarding is another example of a feature that would allow a party to
answer a call that is not the originally called party. The enhancements allow the caller to see the
identity of the answering party for a forwarded call.
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.
8.6.6. Display of Redirection Reason at Forwarded-To Party, Call

Forwarding Between Systems Networked via SIP
This section focuses on the displays that appear on a station receiving a call that has been
forwarded from another Avaya Communication Manager Release 5 system. The SIP Private
Networking enhancements allow the party receiving an incoming call to see the identity of the
originally called party and the associated redirection reason, even when the call forwarding has
occurred on another Avaya Communication Manager Release 5 system networked via SIP
trunks. Like call coverage, call forwarding can also be based on different criteria, such as “busy”
and “no answer”. However, unlike call coverage, the same redirection reason will appear.
Note that the call forwarding features are subject to Class of Service permissions, which are
granted to the users in the sample configuration.

For example, referring to Figure 2, assume John Zack at x58220 activates call forwarding all to
x58430. Assume Janey Digital at x58203 dials local user John Zack at x58220, and the call is
forwarded over the SIP trunk to remote station 58430. The forwarded-to user’s display at
x58430 will be “Janey Digital to John Zack f”. That is, the call history at the remote Avaya
Communication Manager is displayed to the party receiving the forwarded call from the SIP
trunk, and the redirection reason is “f” for “forwarding”.
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.
8.6.7. Display of Connected Party to Originator of Call, Distant Answer via

Bridged Appearance
Bridging is another example of a feature that would allow a party to answer a call that is not the
originally called party. The SIP private networking enhancements allow the caller to see the
identity of the party that has used a bridged appearance to answer the call at a distant Avaya
Communication Manager, when SIP networking is used between systems.
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.
change system-parameters features Page 9 of 17

FEATURE-RELATED SYSTEM PARAMETERS
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.

8.6.8. Display Updates for Call Transfer
Transfer is another example of a feature that can cause a change to a call that is relevant to
displays. In a SIP network, it is possible to receive a display update when a call is transferred at
another Avaya Communication Manager system. For example, assume x58203 dials 58430, and
the call is completed over a SIP trunk. After answering the call, the user at x58430 completes a
transfer to local user “Ed Sobeleone” at x58467 (either blind or attended). The display for
station 58203 updates to “Ed Sobeleone 58467”, where “Ed Sobeleone” is the name associated
with extension 58467. This is an example of a transfer occurring at the distant Avaya
Communication Manager that results in a display update when the inter-system call uses a SIP
trunk.
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

58430. The SIP protocol will carry the privacy indication to the receiving Avaya
Communication Manager system. Assuming no other changes to the configuration, the display
at x58430 will be “Co-Res SES 166”, which is the trunk group name and trunk access code of
the incoming trunk for the call.
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.
change system-parameters features Page 9 of 17

FEATURE-RELATED SYSTEM PARAMETERS
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.
change trunk-group 66 Page 3 of 21

TRUNK FEATURES
ACA Assignment? n Measured: none
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? y

Replace Unavailable Numbers? Y
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”.

If all or most calls that Janey Digital makes should be marked for privacy, the station form for
Janey Digital can be programmed to restrict presentation by default, as shown in bold in the
following screen.

STATION
FEATURE OPTIONS
LWC Reception: spe Auto Select Any Idle Appearance? n
LWC Activation? y Coverage Msg Retrieval? y
LWC Log External Calls? n Auto Answer: none
CDR Privacy? n Data Restriction? n
Redirect Notification? y Idle Appearance Preference? n
Per Button Ring Control? n Bridged Idle Line Preference? n
Bridged Call Alerting? n Restrict Last Appearance? y
Active Station Ringing: single
EMU Login Allowed? n
H.320 Conversion? n Per Station CPN - Send Calling Number? r
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.

http://www.avaya.com/master-usa/en-us/resource/assets/applicationnotes/CoRes-w-DO.pdf
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.
[6] Maintenance and Troubleshooting Guide for Avaya Distributed Office, Document 03-
602029, Issue 1.0, May 2007.
[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].
[8] Installing, Administering, Maintaining, and Troubleshooting SIP Enablement Services,

Document 03-600768, Issue 5.0, Jan 2008.

©2008 Avaya Inc. All Rights Reserved.
Avaya and the Avaya Logo are trademarks of Avaya Inc. All trademarks identified by ® and ™
are registered trademarks or trademarks, respectively, of Avaya Inc. All other trademarks are the
property of their respective owners. The information provided in these Application Notes is
subject to change without notice. The configurations, technical data, and recommendations
provided in these Application Notes are believed to be accurate and dependable, but are
presented without express or implied warranty. Users are responsible for their application of any
products specified in these Application Notes.
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


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Avaya Solution & Interoperability Test Lab

Sample Configuration for SIP Private Networking and SIP

JRR; Reviewed: Solution & Interoperability Test Lab Application Notes 1 of 65

Figure 1: Network Overview Repeated from Reference [1]

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Figure 2: Network Overview

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1.1. Summary of Triggers for SIP Look-Ahead Routing

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• SIP INVITE message can not be transmitted

1.2. Feature Transparency among Networked Independent Systems

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2. Equipment and Software Validated

Network Component Version Information

Table 3 – Equipment and Version Information

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4. Avaya S8500C SES Edge Server Configuration

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4.2. Address Map Configuration

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In the interests of illustrating alternative approaches, in Section 6, Avaya Communication

4.2.1. New Host Address Map for SES Edge

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In the confirmation screen, click Continue.

In the resulting screen shown below, click Add Another Contact.

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The following confirmation screen appears. Click Continue.

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In the Pattern field, the following is entered: “^sip:584[0-9][0-9]@distributed.com”. In the

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5.1. UDP Entry To Reach Users at Avaya S8300B Site

change uniform-dialplan 582 Page 1 of 2

5.2. AAR Entries To Reach Users at Avaya S8300B Site

change aar analysis 2 Page 1 of 2

Dialed Total Route Call Node ANI

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Dialed Total Route Call Node ANI

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change route-pattern 20 Page 1 of 3

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6.1. Node Names

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6.2. SIP Signaling Group and Trunk Group

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Group Number: 2 Group Type: sip

Near-end Node Name: procr Far-end Node Name: SES-Edge

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add trunk-group 2 Page 1 of 21

6.3. Association of Endpoints with Network Regions

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Slot Module Type Name

6.4. Network Region Configuration

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src dst codec direct WAN-BW-limits Video Dyn

6.5. Codec Set Configuration

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Audio Silence Frames Packet