Exchange 2010 Unified Messaging is feature rich and can replace a legacy PBX voicemail system. Avaya communication manager can utilize Unified messaging without a session manager (used to be called Sip enablement server) by interfacing with the Asterisk free pbx.
Software and versions used:
- Avaya Communication Manager 4.0.5
- Asterisk 10.1.2
- Exchange 2010 sp1 version 14.01.0218.013
- opensips 1.7.2
Check your Avaya system to make sure that you have enough Sip trunks licenced.
display capacity
display capacity Page 7 of 12
SYSTEM CAPACITY
System
Used Available Limit
-----------------------
SIP Trunks (included in 'Trunk ports'): 178 222 400
There are 400 sip trunks licenced and 222 available.
Set up a trunk group between Avaya and Asterisk
Avaya Configuration
1. Create node name for remote Asterisk server.
The authoritative domain is set in the ip network region to “avaya.com”
change node-names ip
list node-names Page 4 NODE NAMES Type Name IP Address IP asterisk 10.125.16.115
2. Create Signaling Group
display signaling-group 13 SIGNALING GROUP Group Number: 13 Group Type: sip Transport Method: tcp Near-end Node Name: pbx-ethernet Far-end Node Name: asterisk Near-end Listen Port: 5060 Far-end Listen Port: 5060 Far-end Network Region: Far-end Domain: asterisk Bypass If IP Threshold Exceeded? n DTMF over IP: rtp-payload Direct IP-IP Audio Connections? n IP Audio Hairpinning? n Enable Layer 3 Test? n Session Establishment Timer(min): 3
DTMF over IP: Use rtp-payload (rfc2833)
Near-end Node Name is a c-lan card.
3. Create Trunk Group
display trunk-group 8 Page 1 of 22 TRUNK GROUP Group Number: 8 Group Type: sip CDR Reports: y Group Name: Asterisk COR: 1 TN: 1 TAC: 769 Direction: two-way Outgoing Display? y Dial Access? n Night Service: Queue Length: 0 Service Type: public-ntwrk Auth Code? n Signaling Group: 13 Number of Members: 8
display trunk-group 8 Page 2 of 22 Group Type: sip TRUNK PARAMETERS Unicode Name? y Redirect On OPTIM Failure: 5000 SCCAN? n Digital Loss Group: 18 Preferred Minimum Session Refresh Interval(sec): 600
display trunk-group 8 Page 5 of 22 PROTOCOL VARIATIONS Mark Users as Phone? n Prepend '+' to Calling Number? n Send Transferring Party Information? n Telephone Event Payload Type:
4. Create Uniform Dial Plan to route selected numbers to the Asterisk Server
list uniform-dialplan Page 2 UNIFORM DIAL PLAN TABLE Matching Pattern Len Del Insert Digits Net Conv Node Num 609 4 0 aar n
In this example, 6090 – 6099 will be routed using the aar table.
5. Update AAR table
list aar analysis Page 2 AAR DIGIT ANALYSIS REPORT Dialed Total Route Call Node String Min Max Pattern Type Number 609 4 4 301 pubu
6. Create route pattern
list route-pattern ROUTE PATTERNS Route Name/Trk FRL Hop IXC BCC TSC CA-TSC ITC Service/Feature Pat Pref Grp Lmt 0 1 2 M 4 W Request 301 test sip 1 8 0 user y y y y y n n none rest
display route-pattern 301 Page 1 of 3 Pattern Number: 301 Pattern Name: test 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: 8 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 2: y y y y y n n rest none 3: y y y y y n n rest none 4: y y y y y n n rest none 5: y y y y y n n rest none 6: y y y y y n n rest none
Route pattern 301 sends traffic to the Asterisk box using trunk group 8.
7. Create Hunt Group
display hunt-group 100 Page 1 of 60 HUNT GROUP Group Number: 100 ACD? n Group Name: Asterisk VM Queue? n Group Extension: 6096 Vector? n Group Type: ucd-mia Coverage Path: TN: 1 Night Service Destination: COR: 1 MM Early Answer? n Security Code: Local Agent Preference? n ISDN/SIP Caller Display:
display hunt-group 100 Page 2 of 60 HUNT GROUP Message Center: sip-adjunct Voice Mail Number Voice Mail Handle Routing Digits (e.g., AAR/ARS Access Code) 6096 6096 709
8. Create Coverage Path that will be assigned to stations
display coverage path 32 COVERAGE PATH Coverage Path Number: 32 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: 5 All? n n DND/SAC/Goto Cover? y y Holiday Coverage? n n COVERAGE POINTS Terminate to Coverage Pts. with Bridged Appearances? n Point1: h100 Rng: Point2: Point3: Point4: Point5: Point6:
9. Assign Coverage Path to station
display station 5338 Page 1 of 5 STATION Extension: 5338 Lock Messages? n BCC: 0 Type: 4610 Security Code: * TN: 2 Port: S00532 Coverage Path 1: 32 COR: 1 Name: Chris Coverage Path 2: COS: 15 Hunt-to Station: STATION OPTIONS Time of Day Lock Table: Loss Group: 19 Personalized Ringing Pattern: 1 Message Lamp Ext: 5338 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? y IP Video Softphone? n Customizable Labels? y
display station 5338 Page 2 of 5 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? y Restrict Last Appearance? y Active Station Ringing: single EMU Login Allowed? n H.320 Conversion? n Per Station CPN - Send Calling Number? Service Link Mode: as-needed Multimedia Mode: enhanced Audible Message Waiting? n MWI Served User Type: sip-adjunct Display Client Redirection? n Select Last Used Appearance? n Coverage After Forwarding? s Multimedia Early Answer? n Remote Softphone Emergency Calls: as-on-local Direct IP-IP Audio Connections? n Emergency Location Ext: 5338 Always Use? n IP Audio Hairpinning? n
Asterisk Configuration
1. Create sip trunks to Avaya and to Exchange
sip.conf
[Exchange2010] tcpenable=yes host=10.125.28.5 type=peer nat=no transport=tcp port=5065 context=default dtmfmode=auto disallow=all ; First disallow all codecs allow=ulaw ; Allow codecs in order of preference relaxdtmf=yes canreinvite=no ;this is needed to force asterisk to bridge calls [Avaya-vm] tcpenable=yes host=10.125.15.65 type=peer transport=tcp port=5060 dtmfmode=rfc2833 context=Avaya-vm nat=no canreinvite=no ;this is needed to force asterisk to bridge calls
2. Add extensions for dialing exchange. Additionally, the logic is included to find out if the original dialed number was the voicemail number (6096) or if the avaya coverage path sent the call here. Since my 8YY numbers VDNs are 5 digit and int the 9XXXX range, I also check if the number was dialed from outside.
extensions.conf
[default]
exten => _5XXX,1,Dial(SIP/${EXTEN}@Avaya-vm)
exten => _8XXX,1,Dial(SIP/${EXTEN}@Avaya-vm)
exten => 6095,1,Dial(SIP/${EXTEN}@Exchange2010)
exten => 6096,1,Dial(SIP/${EXTEN}@Exchange2010)
[Avaya-vm]
exten => 6096,1,Answer
exten => 6096,2,Set(orig_exten=${SIP_HEADER(History-Info,3)})
exten => 6096,3,Set(orig_exten=${CUT(orig_exten,<,2)})
exten => 6096,4,GotoIf($[${LEN(${orig_exten})}<1]?22)
exten => 6096,5,Set(orig_exten=${orig_exten:4:4})
exten => 6096,6,GotoIf($["${orig_exten}"=""]?400)
exten => 6096,7,GotoIf($["${orig_exten}"="6096"]?22)
exten => 6096,8,SIPAddHeader(Diversion: <tel:${orig_exten}>;reason=user-busy;screen=no;privacy=off)
exten => 6096,9,Dial(SIP/6096@Exchange2010)
exten => 6096,10,Hangup
exten => 6096,22,Set(orig_exten=${SIP_HEADER(History-Info)})
exten => 6096,23,Set(orig_exten=${orig_exten:5:4})
exten => 6096,24,GotoIf($["${orig_exten}"="6096"]?400)
exten => 6096,25,GotoIf($["${orig_exten}">"8999"]?400)
exten => 6096,26,SIPAddHeader(Diversion: <tel:${orig_exten}>;reason=user-busy;screen=no;privacy=off)
exten => 6096,27,Dial(SIP/6096@Exchange2010)
exten => 6096,400,Dial(SIP/6096@Exchange2010)
;this is the auto attendant number.
exten => 6095,1,Dial(SIP/${EXTEN}@Exchange2010)
Exchange 2010 Configuration
1. Install Exchange Unified Messaging
2. Create New UM Dialplan. Select number of digits in your dialplan and associate with your UM server
3. Under subscriber access put in the voicemail access number.
4.Configure your UM IP Gateway to point to the Asterisk Server.
At this point, almost everything works. The only thing that doesn’t work is the Message Waiting Indicator(MWI) on the Avaya phone.
This is where opensips comes in to fix the issue. In this configuration, Exchange 2010 sends a SIP NOTIFY message to Asterisk with the message-summary.
Opensips Configuration
Here is what the SIP NOTIFY looks like from Exchange to Asterisk
NOTIFY sip:5338@10.125.16.115:5060;user=phone SIP/2.0
FROM: <sip:5338@10.125.16.115:5060;user=phone>;epid=9C9C089B0D;tag=cecc43a247
TO: <sip:5338@10.125.16.115:5060;user=phone>
CSEQ: 5 NOTIFY
CALL-ID: acd683f736af4b4b982822a15eef3ca1
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 10.125.28.5:46028;branch=z9hG4bK5ddc769
CONTACT: <sip:5338@10.125.16.115:5060;user=phone>
CONTENT-LENGTH: 100
EVENT: message-summary
SUBSCRIPTION-STATE: terminated
USER-AGENT: RTCC/3.5.0.0 MSExchangeUM/14.01.0218.012
CONTENT-TYPE: application/simple-message-summary
Messages-Waiting: yes
Message-Account: sip:5338@10.125.16.115:5060;user=phone
Voice-Message: 1/1
The SIP NOTIFY needs to be manipulated and sent to the Avaya PBX.
1. Install Opensips 1.7.2
/etc/opensips/opensips.cfg
Change listening port to 5061
port=5061
if (from_uri=~”.*@10.125.16.115″)
{
replace_all(“CONTENT-LENGTH: 101″,”CONTENT-LENGTH: 81″);
replace_all(“CONTENT-LENGTH: 100″,”CONTENT-LENGTH: 80″);
replace_all(“CONTENT-LENGTH: 99″,”CONTENT-LENGTH: 79″);
replace_all(“10.125.16.115:5061″,”avaya.com”);
replace_all(“;user=phone”,”");
t_relay(“tcp:10.125.15.65:5060″);
}
In exchange management shell run the command:
Set-UMIPGateway -identity Asterisk -Port 5061
This will change the port that Exchange sends the SIP NOTIFY messages to, but will not affect any other communication.
Once this change is made the SIP NOTIFY messages will go to opensips, which will then forward them to the Avaya Communication Manager. The MWI will now work.
