View source for SIP Devices - creating an account

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[[File:ICX SIP Devices.PNG|thumb|right|600px|ICX-AlphaCom as SIP Server]]
The ICX-AlphaCom is hosting an embedded SIP server. This article describes how to '''create a SIP account''' in the ICX-AlphaCom to which SIP devices can register.

Examples of SIP devices are:
* ITSV video intercoms
* VoIP telephones
* PC clients (Softphones)
* WiFi phones
* IP Dect systems
* Analog Telephone Adapter (ATA)

The ICX-AlphaCom supports up to 500 SIP devices.


== ICX-AlphaCom Configuration ==
=== Web Configuration ===
*''' Filter settings''': In [[ICX Web|the ICX-AlphaCom web interface]], go to [[ICX_Web#Filters|System Configuration -> Filters]], and enable '''SIP''' (UDP port 5060) used for SIP signalling, and '''VoIP Audio''' (UDP ports 61000:62000). By default these ports are enabled on Ethernet port 1.
*''' Licenses''': A [[Licenses for ICX-500 and ICX-AlphaCom Core|"SIP station" license (item no 1002600700)]] is required for each SIP device. 
{{note|A license will be consumed as soon as the "SIP Station" flag is enabled in AlphaPro, regardless if the SIP device is registered or not.}}

<br>

=== AlphaPro Configuration===
To create a SIP account, do the following:
*From the '''Users & Stations''' window in AlphaPro, select a free user from the listbox, and enable the '''SIP Station''' flag 
* Configure '''Directory Number''' and '''Display Text'''
* Select a '''Codec''' which is supported by the SIP device
[[Image:ICX SIP Account.PNG|thumb|500px|left|Creating a SIP account for SIP extension 113]]
<br style="clear:both;" />

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== SIP Device configuration ==
Minimum settings in the SIP device:
* The '''IP address of ICX-AlphaCom''' (Typically referred to as SIP Server, Registration Server, Registrar, Domain, Proxy Domain, User Domain)
* The '''Directory number''' as defined in AlphaPro (Typically referred to as SIP User ID, SIP Account, User Name, Register Name). Often the Directory Number also must be entered as SIP Authentication ID.
* Select a '''Codec''' which is matching the codec configured in AlphaPro

A typical SIP device configuration:
[[File:ITSV-4 Accounts.png|thumb|700px|left|Setting up a SIP account in the ITSV-4 video phone]]
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== Feature support ==
SIP devices have some feature limitations compared to native Zenitel IP devices.

=== Feature list===
{| border="1"
! style="background:#ffdead;" width="300" |'''Feature'''
! style="background:#ffdead;" width="170" |'''SIP Devices (incl ITSV)'''
! style="background:#ffdead;" width="170" |'''Zenitel IP Devices'''
|-
| Point to point calls || align=center | <span style="color: green">'''V'''</span> || align=center | <span style="color: green">'''V'''</span> 
|-
| Caller ID || align=center | <span style="color: green">'''V'''</span>  || align=center | <span style="color: green">'''V'''</span>
|-
| [[Duplex Conference]] || align=center | <span style="color: green">'''V'''</span> (4) ||align=center | <span style="color: green">'''V'''</span>
|-
| [[Door opening]] (6) || align=center | <span style="color: green">'''V'''</span>  || align=center | <span style="color: green">'''V'''</span>
|-
| Send [[Call Request]] || align=center | <span style="color: green">'''V'''</span>  || align=center | <span style="color: green">'''V'''</span>
|-
| Member of [[Ringing Group Feature|Ringing Group]] || align=center | <span style="color: green">'''V'''</span> (8) || align=center | <span style="color: green">'''V'''</span>
|-
| Line Down reporting || align=center | <span style="color: green">'''V'''</span> (7)  || align=center | <span style="color: green">'''V'''</span>
|-
| Receive [[Call Request]] (Call Queuing) || align=center | <span style="color: red">'''X'''</span> (6) || align=center | <span style="color: green">'''V'''</span> 
|-
| [[Simplex Conference]] || align=center | <span style="color: red">'''X'''</span>  || align=center | <span style="color: green">'''V'''</span>
|-
| [[Audio Program]] || align=center | <span style="color: red">'''X'''</span>  || align=center | <span style="color: green">'''V'''</span> 
|-
| Receive Group Calls || align=center | <span style="color: green">'''V'''</span>  (1) || align=center | <span style="color: green">'''V'''</span> 
|-
| Receive [[Voice Alarm Messages]] || align=center | <span style="color: green">'''V'''</span>  (1) || align=center | <span style="color: green">'''V'''</span> 
|-
| Receive [[Mail Messages]] (Station errors etc.) || align=center | <span style="color: red">'''X'''</span> || align=center | <span style="color: green">'''V'''</span> 
|-
| [[Call Forward feature|Call Forwarding]] || align=center | <span style="color: red">'''X'''</span> (5) || align=center | <span style="color: green">'''V'''</span> 
|-
| Status info in display (Call Forwarding, Absence status etc.) || align=center | <span style="color: red">'''X'''</span> || align=center | <span style="color: green">'''V'''</span> 
|-
| Volume setting || align=center | <span style="color: green">'''V'''</span> (2) || align=center | <span style="color: green">'''V'''</span> 
|-
|[[Hot-line Dialing|Hotline call]] || align=center | <span style="color: green">'''V'''</span> (3) || align=center | <span style="color: green">'''V'''</span> 
|-
|[[Group hunt|Group Hunt member]] || align=center | <span style="color: red">'''X'''</span> || align=center | <span style="color: green">'''V'''</span> 
|-

| Line Monitoring via [[Line_monitoring#Tone_test|Tone test]] || align=center | <span style="color: red">'''X'''</span> || align=center | <span style="color: green">'''V'''</span> 
|}

*(1) The SIP phone needs to register with a second account (requiring a second SIP phone license). The second account must have AutoAnswer enabled, and be included in the group call. The flag "Allow SIP Stations in Group Calls and as default speaker in SX Conference" must be enabled in [[Exchange_%26_System_(AlphaPro)#VoIP|AlphaPro - Exchange & System > System > VoIP]]
*(2) Volume is set locally from the phone. Volume cannot be controlled from the ICX-AlphaCom server. [[Volume and handset override|Volume Override]] is not supported.
*(3) Must be supported by the phone. The Hot-Line feature in the ICX-AlphaCom cannot be used.
*(4) Only way to join a conference is for the SIP phone to dial the conference directory number. The phone cannot be remotely included in the duplex conference.
*(5) Call Forwarding (71) can be activated on the SIP phone, but there is no status indication in the phone indicating if the Forwarding is active or not. Any forwarding feature within the SIP phone it self is not supported by the ICX-AlphaCom.
*(6) Call Request is not supported on SIP phones, but Ringing Group is supported and is using the same underlying functionality. The difference is that the calls are queued in the server, not in the phone. So the user will not see any queue. As soon as the first call is terminated, the next call will come through.
*(7) Line down reporting on SIP phones is based on the "Registration expire" time out. This timeout is default quite long (30 minutes or more) on most SIP phones. If faster fault reporting is needed the registration timeout must be adjusted in the phone.
*(8) Calls via the Ringing Group feature are queued ''in the server'', not in the phone. So the user will not see the call queue. As soon as the first call is terminated, the next call will come through.



===Line Down reporting ===
Line down reporting on SIP phones is based on the "Registration expire" time out. 
This timeout is default quite long (30 minutes or more) on most SIP phones. If faster fault reporting is needed the registration timeout must be adjusted in the phone.

== Related articles ==
* [[SIP_trunk_node_-_configuration#SIP_Trunk_Station|SIP Trunk Station]]

[[Category: ICX-AlphaCom - SIP Integration]]