Interoperability

  1. SIP Phones
    1. How to set paging function on the phone side
    2. SIP softphone we recommend
    3. Hard Phones
      1. ClearOne MAX IP Conferencing Phone
      2. Grandstream Networks Budge Tone - 100 Phone
      3. inter-tel
      4. ipDialog SIP Tone III Phone
      5. ipDialog SipTone™ V Phone
      6. Linksys IP Phone
      7. Nokia Phone
      8. Panasonic KX-TGP550 T04 DECT
      9. Panasonic KX-UT1xx
        1. Configure shared line in Panasonic KX-UT1xx
        2. Set up Flexible Button on Panasonic KX-UT1xx
      10. Polycom Kirk DECT
      11. Polycom SoundPoint IP 650/330 Phone
      12. Polycom SpectraLink 8020 Wireless Telephone
      13. Polycom VVX600 Business Media Phone
      14. SpectraLink 8400 Wireless Telephone
      15. SAXA IP Netphone SX Phone
      16. Snom m3 VoIP wireless DECT phone
      17. Snom Phones
      18. Swissvoice IP10S Phone
      19. Ubiquiti UniFi VoIP Phones
      20. Vtech Business Phones
      21. Yealink SIP Phones
      22. ZyXEL Prestige 200w
    4. Soft Phones
      1. Age Phone
      2. Bria
      3. Kapanga
      4. X-Lite
  2. VoIP SIP Gateways
    1. AddPac AP1200
    2. AudioCodes Mediant 1000(FXO/FXS)
    3. AudioCodes MP-104 (FXO)
    4. AudioCodes MP-114 (FXO/FXS)
    5. AudioCodecs TP-260
    6. Cisco Systems ATA186
    7. Grandstream Networks HondyTone 486
    8. Mediatrix 1104(FXS)
    9. Linksys SPA-3000
    10. Mediatrix 1204 (FXO)
    11. Medicatrix 2102 (FXS)
    12. Quintum Tenor ASM200
    13. Quintum Tenor DX Gateway
    14. Sangoma NetBorder Express Gateway
    15. TAINET - Venus 2908
    16. Yamaha RTV 700
    17. Yamaha RTX 1000
    18. ZyXEL Prestige 2602 HW
  3. ITSP
    1. BroadVoice
    2. DIDx
    3. iConnectHere
    4. InPhonex
    5. Megapath
    6. sipgate
    7. Skype
    8. Triad Telecom
    9. VoIP INNOVATIONS
    10. Voxbone
    11. VoIPUSER
    12. Vonage
  4. WebRTC client
    1. JsSIP
  5. Misc
    1. Application-Level Gateway(ALG)
      1. InGate SIParator 4.6.1
        1. Standalone SIParator, with SIP server on the outside
        2. DMZ SIParator, with Brekeke SIP Server on the outside
    2. Ascom AA60
    3. Avaya PBX (CM 5.2)
    4. CyberData VoIP Ceiling Speaker
    5. CyberData VoIP Intercom
    6. DTH VoIP Billing
    7. Microsoft Lync
    8. Microsoft Lync or OCS
      1. Brekeke SIP Server and Lync or OCS
      2. Brekeke PBX and Microsoft Lync or OCS
        1. Configure Brekeke PBX for Lync or OCS
        2. Configure Lync or OCS for Brekeke PBX
    9. Microsoft Speech Server
    10. Mobotix T24M-Sec-D11 Hemispheric IP Video Door Station
    11. Nortel CS1000
    12. Radius Server
      1. Jerasoft Development - JeraSoft VCS
      2. Jerasoft Development's VCS Dynamic Routing
      3. Clearbox Radius
      4. FreeRADIUS
    13. Jeron's Nurse Call System - Provider 790
    14. Rauland's Nurse Call System - Responder 5
    15. Sangoma NetBorder Call Analyzer

Configure Brekeke SIP Server with Avaya PBX (CM 5.2)

Brekeke SIP Server Setup

The following DialPlan rules can allow you to make calls between Brekeke SIP Server registered UA and Avaya PBX

Rule-1: From Avaya
----------------------------------------------- 
[Matching Patterns]
$request = ^INVITE
$addr = <Avaya_PBX_Address>
To = sip:(.+)@

[Deploy Patterns]
To = sip:%1@
$auth = false
-----------------------------------------------

With the rule “From Avaya”, Brekeke SIP Server can accept call from Avaya PBX without checking authentication and send call to related UA registered at Brekeke SIP Server.

Rule-2: To Avaya
----------------------------------------------- 
[Matching Patterns]
$request = ^INVITE
$registered = false
To = sip:(.+)@

[Deploy Patterns]
$auth = false 
To = sip:%1@<Avaya_PBX_Address>
$transport = tcp
-----------------------------------------------

With the rule “To Avaya”, when the call is to an unregistered number, Brekeke SIP Server will send call to Avaya’s IP Address with TCP transport.

Avaya side setup

Create trunk group, Signalling group as given bleow:

**************************TRUNK - ~START**************
Trunk group details:
display trunk-group 101 Page 1 of 21
TRUNK GROUP

Group Number: 101 Group Type: sip CDR Reports: y
Group Name: BRKEKE SIP COR: 1 TN: 1 TAC: 186
Direction: two-way Outgoing Display? n
Dial Access? n Night Service:
Queue Length: 0
Service Type: public-ntwrk Auth Code? n
Signaling Group: 101
Number of Members: 5

display trunk-group 101 Page 2 of 21
Group Type: sip

TRUNK PARAMETERS
Unicode Name: yes
Redirect On OPTIM Failure: 5000
SCCAN? n Digital Loss Group: 18
Preferred Minimum Session Refresh Interval(sec): 64800 or 90

display trunk-group 101 Page 3 of 21
TRUNK FEATURES
ACA Assignment? n Measured: both
Maintenance Tests? y
Numbering Format: public
UUI Treatment: service-provider
Replace Restricted Numbers? n
Replace Unavailable Numbers? n
Show ANSWERED BY on Display? y

display trunk-group 101 Page 4 of 21
PROTOCOL VARIATIONS
Mark Users as Phone? n
Prepend '+' to Calling Number? n
Send Transferring Party Information? n
Network Call Redirection? n
Send Diversion Header? n
Support Request History? y
Telephone Event Payload Type: 127

display trunk-group 101 Page 5 of 21
TRUNK GROUP
Administered Members (min/max): 1/5
GROUP MEMBER ASSIGNMENTS Total Administered Members: 5
Port Name
1: T00416 BRKEKE SIP
2: T00417 BRKEKE SIP
3: T00418 BRKEKE SIP
4: T00419 BRKEKE SIP
5: T00420 BRKEKE SIP
6:
7:
***************END***************************
************Signalling group START********************
The SIGNALLING group details are as given below:

display signaling-group 101
SIGNALING GROUP

Group Number: 101 Group Type: sip
Transport Method: tcp
IMS Enabled? n

Near-end Node Name: <Near-End-Node_Name> Far-end Node Name: 
Near-end Listen Port: 5060 Far-end Listen Port: 5060
Far-end Network Region: 100
Far-end Domain: 

Bypass If IP Threshold Exceeded? n
Incoming Dialog Loopbacks: eliminate RFC 3389 Comfort Noise? n
DTMF over IP: rtp-payload Direct IP-IP Audio Connections? n
Session Establishment Timer(min): 3 IP Audio Hairpinning? y
Enable Layer 3 Test? n Direct IP-IP Early Media? n
H.323 Station Outgoing Direct Media? n Alternate Route Timer(sec): 6

***************end************
*******Avaya IP NETWORK REGION START************

display ip-network-region 100 Page 1 of 19
IP NETWORK REGION
Region: 100
Location: 1 Authoritative Domain: 
Name: SIP BREKEKE
MEDIA PARAMETERS Intra-region IP-IP Direct Audio: yes
Codec Set: 6 Inter-region IP-IP Direct Audio: yes
UDP Port Min: 2048 IP Audio Hairpinning? n
UDP Port Max: 65531
DIFFSERV/TOS PARAMETERS RTCP Reporting Enabled? y
Call Control PHB Value: 26 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: 3
Audio 802.1p Priority: 5
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

IP NETWORK REGION

INTER-GATEWAY ALTERNATE ROUTING / DIAL PLAN TRANSPARENCY
Incoming LDN Extension:
Conversion To Full Public Number - Delete: Insert:
Maximum Number of Trunks to Use for IGAR:
Dial Plan Transparency in Survivable Mode? n

BACKUP SERVERS(IN PRIORITY ORDER) H.323 SECURITY PROFILES
1 1 challenge
2 2
3 3
4 4
5
6 Allow SIP URI Conversion? y

TCP SIGNALING LINK ESTABLISHMENT FOR AVAYA H.323 ENDPOINTS
Near End Establishes TCP Signaling Socket? y
Near End TCP Port Min: 61440
Near End TCP Port Max: 61444

display ip-network-region 100 Page 3 of 19

Source Region: 100 Inter Network Region Connection Management I M
G A e
dst codec direct WAN-BW-limits Video Intervening Dyn A G a
rgn set WAN Units Total Norm Prio Shr Regions CAC R L s
1 1 y NoLimit n
2 1 y NoLimit n
3 1 y NoLimit n

display ip-network-region 100 Page 9 of 19

Source Region: 100 Inter Network Region Connection Management I M
G A e
dst codec direct WAN-BW-limits Video Intervening Dyn A G a
rgn set WAN Units Total Norm Prio Shr Regions CAC R L s
91
92
93
94
95
96
97
98
99
100 6 all
101 6 y NoLimit n all
102
103
104
105
************END*****************

With the above settings, you can connect the Brekeke SIP Server with AVAYA Communication Manager.

Yes No
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