Administrator Guide: Communication Server 2

Table of Contents

Goal of the document and intended audience

Changes since the Asterisk-1.2x module

Global pickup

 Navigate to: SMP > Communication Routing > Routes

• Click add
• Telephony route: Choose anything you like. This is the number you will dial on the phone. The one in the Asterisk-1.2 module was *9.
• Description: Global Pickup
• Route group: Choose a route group that is included in the restriction group of your phones. Typically, national or service are included.
• Action: MapNumber.1.16 or higher
• Click save

• Click on the MapNumber action for your new route
• Manipulate number: Strip all digits
• Add prefix: The root extension of the callflow. The one in the example is *210.
• Click save
• Apply changes

Zeroconf - Allow Guest Calls

No support for Digium hardware

Queue preferred device

SIP overload

unavailable parameters

• CALLERIDNUM is no longer available. You can use CALLERID(num) instead.
• CALLERIDNAME is no longer available. You can use CALLERID(name) instead.

Dependencies and minimal required versions

Dependencies

Software and hardware

Mandatory Modules

• Database Server Module >= 2.2.2
• Dahdi Module >= 1.0.0
• Music On Hold Module >= 1.0.0
• SOP Base Module >= 1.4.7
• Sounds Module >= 1.8.2

Mandatory Resources

• MusicOnHold >= 1.6
• MusicOnHoldSelector >= 1.1

Minimal versions of optional modules

• SOP API >= 4.0.2, but SOP API >= 4.1.1 is recommended
• Application Management Server >= 3.3.4
• CallManagement Module >= 1.6.1
• Cisco Phone Support >= 4.2.0
• net.Desktop Module >= 2.26.0 but net.Desktop >= 2.27.0 is recommended
• net.Console Module >= 3.4.1
• net.Supervisor Module >= 1.4.3
• PUM >= 3.2.3
• SNMP Agent Module >= 2.7.5
• Sangoma Card Support Module >= 2.3.0
• SNOM Phone Support >= 1.13.0
• VMXML >= 1.3.1

Minimal versions of resources and resource compatibility

Validated resources

Phone resources

• All Polycom resources >= 4.16.0
• All Snom 3x0 resources >= 3.10.0
• Snom M3 - SDO9 >= 1.4
• Snom 870 - SDO7 >= 3.11
• Aastra 610d >= 1.2
• Aastra 620d >= 1.2
• All other Aastra resource >= 1.06
• Cisco SPA2102 >= 1.13.1
• Cisco SPA3102 >= 1.3.0
• Cisco SPA8000 >= 1.6.0
• Cisco SPA112 >= 1.13.1
• Cisco7905 >= 1.04
• Cisco7912 >= 1.06
• Cisco7940 >= 2.13
• Cisco7941 >= 1.1.0
• Cisco7942 >= 1.2.0
• Cisco7960 >= 2.2.0
• Grandstream GXP2000 >= 2.4
• SangomaAnalogInterface >= 1.2.0
• GenericSIPPhone >= 1.05
• EyebeamAudio >= 1.20
• Grandstream Virtual Phone >= 1.7
• Polycom Virtual Phone >= 1.09
• Aastra Virtual Phone >= 1.04
• UnidataWireless7700 - SDU1 >= 1.8
• UnidataWireless7800 - SDU2 >= 1.2

Interface resources

• OutgoingIAXTrunk >= 1.13
• EndPointAbstractionInterface >= 1.40
• SIP Trunk >= 1.43.1
• MeshSIPTrunk >= 1.12.1
• SangomaBRI >= 1.1.0
• SangomaPRI >= 1.2.0

MusicOnHold resources

• MusicOnHold >= 1.6
• MusicOnHoldSelector >= 1.1

Desktop Application resources

• net.Supervisor >= 1.1
• net.Desktop X100 >= 1.3
• net.Desktop X300 >= 1.6
• net.Desktop X350 >= 1.1
• net.Desktop X500 >= 1.11
• net.Console X700 >= 1.4
• net.Console X900 >= 1.6

Other resources

• Queue >= 2.2

Incompatible resources

Phone resources

• AnalogInterface - ZDA1
• Snom 820 - SDO5

Interface resources

• Zaptel PRI Trunk (previously OutgoingPRITrunk)
• CAPI Trunk (previously OutgoingBRITrunk)
• OutgoingEnumLookup
• mISDN Trunk (previously OutgoingMISDNTrunk)
• OutgoingZapAnalogTrunk
• WoomeraTrunk

MusicOnHold resources

• MusicOnHold < 1.6
• MusicOnHoldSelector < 1.1

Unvalidated resources

Phone resources

• Mobile Phone - LDA1
• IdefiskFree - IDI1
• HitachiWireless3000 - SDA1
• HitachiWireless5000 - SDA2
• HitachiWireless8000 - SDA3
• BudgetTone102 - SDB2
• HandyTone - SDH1
• All Mitel models - SDM1 -> SDM5
• Snom 821 - SDO6
• Snom M9 - SDOA
• ThomsonST2022 - SDT1
• ThomsonST2030 - SDT2
• SwissVoiceIP10S - SDW1

Minimal versions of actions and action compatibility

Validated actions

• AddQueueMember >= 1.2
• AddToConference >= 1.6
• Answer >= 1.1
• AutoRecall >= 2.0.0
• Busy >= 1.05
• CallDevices >= 1.25
• CallDevicesByExtension >= 1.12
• CallExtensions >= 1.6
• CallInterface >= 1.7
• CallService >= 1.3
• CallUsersByGroup >= 1.22
• CheckAndUpdateKey >= 1.1.0
• CheckDate >= 1.4
• CheckDeviceAvailablity >= 1.2
• CheckHoliday >= 1.1
• CheckPincode >= 1.2
• CheckResourceGroupAvailability >= 1.3
• Congestion >= 1.01
• DialOnDevices >= 1.11.0
• DirectedCallPark >= 1.1
• Directory >= 1.0
• DISA >= 1.3.0
• ExtendedCallDevices >= 1.21.0
• Foreach >= 1.1
• GetCallInfo >= 1.1
• GetDigit >= 1.5
• GetEndpointAbstraction >= 1.1.0
• GetExtensionInfo >= 1.3
• GetExtensions >= 1.5
• GetLastCdr >= 1.0.0
• GetResourceInfo >= 1.1.0
• GetSiteFromIP >= 1.1
• GetXpath >= 1.4
• Goto.APPLICATION >= 1.3
• Goto.DefaultOut >= 1.1
• Goto.INTERFACE >= 1.40.1
• Hangup >= 1.2
• If >= 1.6.0
• Intrude >= 1.2.0
• IVR >= 1.5
• MapDDI >= 3.1.0
• MapNumber >= 1.16
• Milliwatt >= 1.0
• PageDevices >= 1.1.0
• Park >= 1.1.0
• PauseQueueMember >= 1.1.0
• Pickup >= 2.2
• PlayPrompt >= 1.3
• PlayText >= 1.1.0
• PlayVariable >= 1.1
• Queue >= 1.6
• ReceiveIM >= 1.1
• RecordCall >= 1.2
• RecordPrompt >= 1.3.1
• Redirect >= 1.3
• RemoveQueueMember >= 1.1
• Ring >= 1.1.0
• SendIM >= 1.1
• SendMail >= 1.1
• SetAccountCode >= 1.1
• SetCallerID >= 1.7.1
• SetChannelGroup >= 1.2.0
• SetLanguage >= 1.2.0
• SetMusicOnHold >= 1.0.0
• SetRingTone >= 1.21.3
• SetVar >= 1.02
• STARTAPPLICATION >= 3.00
• STARTDYNAMICAPPLICATION >= 7.1
• STARTFAXAPPLICATION >= 3.10
• STARTRECURRENTAPPLICATION >= 1.03
• Supervision >= 1.3
• Switch >= 1.0
• Transfer >= 1.1.0
• UnPauseQueueMember >= 1.1.0
• Voicemail >= 1.11
• VoicemailMenu >= 1.4
• Wait >= 1.3

Incompatible actions

• CheckMediaLinkCongestion. Call admission control is built into the Communication Server module.
• ExecuteCallflow. Deprecated in favor of MapNumber
• Goto.BRI. Deprecated in favor of Goto.INTERFACE
• Goto.PRI. Deprecated in favor of Goto.INTERFACE
• InitService. Deprecated in favor of STARTDYNAMICAPPLICATION >= 7.0
• ProxyCall. Deprecated
• RequestInterfaceCallback. Will not be supported
• ResumeCall. Deprecated.

Upgrading from the asterisk-1.2x module

• Are you running a supported baseline installation? You need at least baseline 2.0.0 before you can install the Communication Server module. Installation on older baselines will fail!
• Are you running a supported SMP version? You need at least SMP version 4.8.1. The module will install correctly on lower versions, but functionality like "Apply changes" might not work on them.
• Are the actions used in your callflows compatible with the Communication Server module? Review the list above to make sure they are. You can automate this by using a specially created task just for this purpose: MigrateActionsToCommunicationServer. This task can upgrade all the actions in your callflows and gives recommendations about actions that might not be compatible at all.
• Are the resources defined in the SMP compatible with the Communication Server module? Review the list above to make sure they are.
• Have you installed all module dependencies? Verify the list above to make sure they are all installed.

Downgrading back to the asterisk-1.2x module

Configuring support for T.38

Configuring Call Admission Control

Configuring Connected Line Identification Presentation

Multi codecs and SIP re-invite support

• If SIP re-invite is enabled everywhere (SIP peers, SIP trunks, Communication Server), multi codecs environments are not supported.

• If a multi codecs environment is needed, SIP re-invite must be disabled on all MeshSipTrunk interfaces to avoid double re-invite.