This is product of my work, researching Avaya features “on the fly”, it could become into a faq in the future.
Necessary elements
An IPO central unit (406, 412, 500, SOE)
VCM expansion card installed: SOE has it by default.
An IP trunk license, it can be a trial one.
A SIP ITSP (internet telephony service provider): some of them give free trial calls.
Where do i Start from, and step by step guide
First select your SIP ITSP: I suggest voipcheap.co.uk because it’s free trial calls that may help you to do plenty of tests without using your money.
(some other suggestions for proven providers may be written here)
Get your username and password, and get the configuration parameters.
Enter your Avaya IPO Manager.
Go to Line | SIP Line
Create new “SIP Line”
Now you have to configure the IP trunk (not all parameters have to be filled)
Line number: it’s the number to identify the trunk inside the IPO but you can assign it freely
ITSP domain name: voipcheap.co.uk
ITSP IP address: you can leave it blank (it’s stated in the manual)
Primary authentication name: your ITSP user name
Primary authentication password: your password
Primary Registration Expiry: Default = 3600 minutes.
This setting defines how often registration with the SIP ITSP is required following any previous registration. (untouched)
Secondary Authentication Name
(untouched)
Secondary Authentication Password: (untouched)
Secondary Registration Expiry: Default = 3600 minutes. (untouched)
Registration Required: Default = Off. If selected, the SIP trunk will register with the ITSP using the value in the ITSP Domain Name field. I switched to ON
In Service: Default = On. LEAVE IT ON, it’s similar to the on/off on analog trunks.
Use Tel URI: Default = Off. (untouched) Voipcheap.co.uk doesn’t need it as far I know
VoIP Silence Suppression: Default = Off. Leave it off, and after some calls try to switch it on and test results.
RE-INVITE Supported: Default = Off. (untouched)
Compression Mode selected. Leave it at default, and after some calls try to switch it to other codecs and test results. In the voipcheap website It says that supported codes are: G.711 (64 kbps) - G.726 (32 kbps) - G.729 (8 kbps) - G.723 (5.3 & 6.3 kbps) and
GSMFR (13.2 kbps) is Temporarily unavailable due to technical difficulties.
And now the network part…
Network Configuration
Layer 4 Protocol: Default = UDP, leave it as SIP, since SIP is the mostly used protocol for all the World of VOIP, but TCP SIP implementations are used nowadays.
Send Port: Default = 5060
Listen Port: Default = 5060
Use Network Topology Info: Default = LAN1
My particular network config for the IPO SOE, which is acting as PBX and as network switch are the following:
System | LAN1
System | LAN1 | Lan Settings
IP Address: Default = 192.168.42.1
IP Mask: Default = 255.255.255.0
Primary Trans. IP Address: Default = 0.0.0.0 (Disabled)
RIP Mode: Default = None – the ipo is not making any routing at all
Enable NAT: Default = Off
DHCP Mode: enabled for 16 clients
(HERE IS THE MISTERY, HELP ME TO DE-MISTIFY)
System | LAN1 | Gatekeeper
It’s a new feature of the 4.0 release, formerly know as H.323 Gatekeeper since it was the only codec family supported for VoIP.
Remember that a gatekeeper is like a “virtual” or switch that which translate network addresses and aliases to make connections via the H.323 protocol trough the data network, it also manages the number of allowed VoIP connections.
H323 Gatekeeper Enable: Default = On
SIP Proxy Enabled: Default = On
This settings enables support of SIP trunks. It also requires entry of a SIP Trunk Channels license.
Note that SIP trunks are NOT supported on the Small Office Edition control unit and IP500 control units running in IP Office Lite mode.
System | LAN1 | Network Topology Settings
It says that this settings can be automatically discovered, but you can configure it manually.
This section is mostly unknown for me, but I can explain some aspects.
STUN Server IP Address: Default = 69.90.168.13
This address is part of “Peer1 Network Inc.”
Run STUN: This Test the connection, and auto config some items
Firewall/NAT Type: Default = Unknown (untouched)
Binding Refresh Timeout: Default = 0 (Never)
Public IP Address: Default = 0.0.0.0
The value provided by the SIP ITSP or discovered by the Run STUN process.
In this case voipcheap provides stun.voipcheap.co.uk, port 3478, there’s not a numeric ip address but it you ping it you get 192.221.62.109
About SIP URI, still untouched, but you can get further info on RFC 3261
Troubleshooting
The first thing before al lis this warning, found at IPO manager (of 4.0) help file that states Note that SIP trunks are NOT supported on the Small Office Edition control unit and IP500 control units running in IP Office Lite mode.
I’m in the process of request an IPO license for mi little big SOE, but it seems that it will be useless, anyway I will be using the SOE to make test VOIP connections.
Necessary elements
An IPO central unit (406, 412, 500, SOE)
VCM expansion card installed: SOE has it by default.
An IP trunk license, it can be a trial one.
A SIP ITSP (internet telephony service provider): some of them give free trial calls.
Where do i Start from, and step by step guide
First select your SIP ITSP: I suggest voipcheap.co.uk because it’s free trial calls that may help you to do plenty of tests without using your money.
(some other suggestions for proven providers may be written here)
Get your username and password, and get the configuration parameters.
Enter your Avaya IPO Manager.
Go to Line | SIP Line
Create new “SIP Line”
Now you have to configure the IP trunk (not all parameters have to be filled)
Line number: it’s the number to identify the trunk inside the IPO but you can assign it freely
ITSP domain name: voipcheap.co.uk
ITSP IP address: you can leave it blank (it’s stated in the manual)
Primary authentication name: your ITSP user name
Primary authentication password: your password
Primary Registration Expiry: Default = 3600 minutes.
This setting defines how often registration with the SIP ITSP is required following any previous registration. (untouched)
Secondary Authentication Name
(untouched)
Secondary Authentication Password: (untouched)
Secondary Registration Expiry: Default = 3600 minutes. (untouched)
Registration Required: Default = Off. If selected, the SIP trunk will register with the ITSP using the value in the ITSP Domain Name field. I switched to ON
In Service: Default = On. LEAVE IT ON, it’s similar to the on/off on analog trunks.
Use Tel URI: Default = Off. (untouched) Voipcheap.co.uk doesn’t need it as far I know
VoIP Silence Suppression: Default = Off. Leave it off, and after some calls try to switch it on and test results.
RE-INVITE Supported: Default = Off. (untouched)
Compression Mode selected. Leave it at default, and after some calls try to switch it to other codecs and test results. In the voipcheap website It says that supported codes are: G.711 (64 kbps) - G.726 (32 kbps) - G.729 (8 kbps) - G.723 (5.3 & 6.3 kbps) and
GSMFR (13.2 kbps) is Temporarily unavailable due to technical difficulties.
And now the network part…
Network Configuration
Layer 4 Protocol: Default = UDP, leave it as SIP, since SIP is the mostly used protocol for all the World of VOIP, but TCP SIP implementations are used nowadays.
Send Port: Default = 5060
Listen Port: Default = 5060
Use Network Topology Info: Default = LAN1
My particular network config for the IPO SOE, which is acting as PBX and as network switch are the following:
System | LAN1
System | LAN1 | Lan Settings
IP Address: Default = 192.168.42.1
IP Mask: Default = 255.255.255.0
Primary Trans. IP Address: Default = 0.0.0.0 (Disabled)
RIP Mode: Default = None – the ipo is not making any routing at all
Enable NAT: Default = Off
DHCP Mode: enabled for 16 clients
(HERE IS THE MISTERY, HELP ME TO DE-MISTIFY)
System | LAN1 | Gatekeeper
It’s a new feature of the 4.0 release, formerly know as H.323 Gatekeeper since it was the only codec family supported for VoIP.
Remember that a gatekeeper is like a “virtual” or switch that which translate network addresses and aliases to make connections via the H.323 protocol trough the data network, it also manages the number of allowed VoIP connections.
H323 Gatekeeper Enable: Default = On
SIP Proxy Enabled: Default = On
This settings enables support of SIP trunks. It also requires entry of a SIP Trunk Channels license.
Note that SIP trunks are NOT supported on the Small Office Edition control unit and IP500 control units running in IP Office Lite mode.
System | LAN1 | Network Topology Settings
It says that this settings can be automatically discovered, but you can configure it manually.
This section is mostly unknown for me, but I can explain some aspects.
STUN Server IP Address: Default = 69.90.168.13
This address is part of “Peer1 Network Inc.”
Run STUN: This Test the connection, and auto config some items
Firewall/NAT Type: Default = Unknown (untouched)
Binding Refresh Timeout: Default = 0 (Never)
Public IP Address: Default = 0.0.0.0
The value provided by the SIP ITSP or discovered by the Run STUN process.
In this case voipcheap provides stun.voipcheap.co.uk, port 3478, there’s not a numeric ip address but it you ping it you get 192.221.62.109
About SIP URI, still untouched, but you can get further info on RFC 3261
Troubleshooting
The first thing before al lis this warning, found at IPO manager (of 4.0) help file that states Note that SIP trunks are NOT supported on the Small Office Edition control unit and IP500 control units running in IP Office Lite mode.
I’m in the process of request an IPO license for mi little big SOE, but it seems that it will be useless, anyway I will be using the SOE to make test VOIP connections.
