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How Calls Are Processed
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5. Under Routing, create Route 3 and do the following:
Enable both SIP peers as source peers.
Enable the first PSTN interface as a Master destination.
Confirm with OK.
6. Save the configuration in the main configuration interface.
If calls other than those beginning with 1 or 2 should also be directed to one peer, remove
the condition from the respective PSTN to SIP route and move the route to the end of the
list.
Load Balancing or Failover between Two SIP Peers
To configure two servers for load balancing or failover, follow these steps:
1. Under PSTN Interfaces, enable and configure all required PSTN interfaces. Confirm
with OK.
2. Under SIP Peers, create both SIP peers and make sure the entry in Domain exactly
matches the domain used by the SIP peer in its SIP address for outgoing calls. Do
not enable the option Default peer for received SIP calls for any of these peers.
If you configure a failover, SIP peer 1 (a Master) should have the option Alive
check enabled. Confirm with OK.
3. Under Routing, create Route 1 and do the following:
Enable the first PSTN interface as a source peer.
Enable the first SIP peer configured in Step 2 as a Master destination. For
load-balancing configurations, SIP peer 2 should be configured as a Master
destination. For failover configurations, it should be configured as a Slave
destination.
Confirm with OK.
4. Under Routing, create Route 2 and do the following:
Enable both SIP peers as source peers.
Enable the first PSTN interfaces as a Master destination.
Confirm with OK.
5. Save the configuration in the main configuration interface.
6. Under Routing, create Route 2 and repeat Step 3 for the second SIP peer with the
only difference that the called address condition for this route is 2.*.
7. Under Routing, create Route 3 and do the following:
Enable both SIP peers as source peers.
Enable the first PSTN interface as a Master destination.
Confirm with OK.
8. Save the configuration in the main configuration interface.
If calls other than those beginning with 1 or 2 should also be directed to one peer, remove
the condition from the respective PSTN to SIP route and move the route to the end of the
list.
- Dialogic 1
- Copyright and Legal Notice 3
- Table of Contents 5
- 1. Introduction 9
- General Features 11
- Call Handling Features 12
- Media Processing Features 13
- Supported RFCs 13
- Enhanced Routing 14
- DSP-Based Features 16
- Fax and Modem Features 16
- Q.SIG Features 17
- PBX Interoperability 18
- Preparing for Configuration 19
- License Activation 19
- Device Unique ID (DUID) 20
- Proof of Purchase Code (PPC) 20
- To Activate a License File 22
- The Board Monitor 25
- The View Report Option 26
- Public Line ISDN Protocols 30
- Carrier Grade 31
- PBX Protocols 31
- Specific Major PBX Types 31
- Configuration Tips and Hints 34
- PSTN Interfaces 39
- Enhanced 40
- Address Normalization 42
- Network Interfaces 45
- SIP Peers 46
- Security 51
- Session Timer 52
- Authentication 54
- Routing 55
- Dialplan) 56
- Conditions 57
- Address Manipulation 60
- Security Profiles 61
- Global Security Parameters 62
- Diva SIPcontrol 64
- Use Case for LDAP 64
- LDAP Query 65
- LDAP Domain 66
- LDAP Cache 68
- Dialplans 69
- Address Maps 72
- Cause Code Maps 76
- Codec Profiles 78
- Registrations 80
- Logging and Diagnostics 82
- 6. Data Security 83
- Secure RTP 84
- SIPcontrol 85
- 7. How Calls are Processed 93
- Emergency Calls 96
- Routing Conditions 96
- Routing Examples 96
- Common Expressions: 103
- Common Results 103
- Address Map Examples 104
- 9. Software Uninstallation 107
- 10. Cause Code Mapping 109
- 11. Event Logging 121
- Warnings 122
- Informational Messages 123
- 12. Use Case Examples 125
- Microsoft® Lync™ Server 2010 175
- 13. SNMP Support 199
- SNMP Support 201
- Configuration Manager 203
- Diva® Line Test Tool 211
- Performing a Line Check Test 211
- Performing a Hardware Test 213
- Performing a Phone/Loop Test 213
- Advanced setup 214
- Advanced transfer setup 215
- Performing a Fax Test 216
- Diagnostics Tool 219
- 17. Customer Service 225
- Dialogic Customer Support 226
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