innovaphone wiki - User contributions [en]

Howto:Power Consumption

2022-10-07T14:16:43Z

Cb: /* Use of extension modules */

==Supply via Power over Ethernet (PoE)==
===Power level===
The power consumption of PoE devices is separated into several classes for power sourcing equipment (PSE) and the powered device (PD).
{| border=1
|+
|Class
|Usage
|Power level PSE [W]
|Power level PD [W]
|-
|0
|default
|15,4
|0,44 - 12,95
|-
|1
|optional
|4
|0,44 - 3,84
|-
|2
|optional
|7
|3,84 - 6,49
|-
|3
|optional
|15,4
|6,49 - 12,95
|-
|4
|optional
|30
|12,95-25,5 W (802.3at/Typ 2)
|}

===Power level of innovaphone devices===
Gateways:
{| border=1
|+
|Device
|PoE-Class
|average total input power [mW]*
|max total input power [mW]*
|-
|IP22
|3
|3000
|4600
|-
|IP24
|3
|3600
|6800
|-
|IP28
|3
|4000
|12900
|-
|IP38
|1
|2800
|3800
|-
|IP302
|3
|3000
|4800
|-
|IP305
|2
|2900
|3000
|-
|IP800 up to HW201
|0
|
|
|-
|IP800 ex HW300
|3
|4700
|10500
|-
|IP2000
|3
|5900
|6500
|-
|IP6000
|3
|9800
|10000
|-
|IP1200
|0
|5000
|8000
|-
|IP1202/IP1203
|2
|
|5000
|-
|IP0010
|3
|8000
|12500
|-
|IP3010
|3
|8000
|12500
|-
|IP6010
|3
|8000
|12500
|-
|IP1060
|3
|8000
|12500
|-
|IP810
|3
|8000
|12500
|-
|IP311/IP411/IP811/IP29/IP0011
|3
|5000
|12500
|-
|IP1130/IP3011
|3
|8000
|12500
|-
|IP29-2/4/8
|3
|May vary in dependency of FXS usage.
|12500
|-
|IP29-20
|4
|May vary in dependency of FXS usage.
|25500
|-
|IP6013
|4
|n.a.
|25500
|-
|}

Phones:
{| border=1
|+
|Device
|PoE-Class
|idle state [mW]****
|average total input power [mW]*
|max total input power [mW]*
|manufacturing check [mW]**
|-
|IP110
|1
|
|2500
|2700
|3500
|-
|IP200A
|1
|
|2900
|3200
|3500
|-
|IP230
|2
|
|2600
|2900
|3500
|-
|IP240
|2
|
|3100
|3800
|4000
|-
|IP150
|1
|
|2800
|3000
|3500
|-
|IP240-1000
|2
|
|4900
|5900
|6500
|-
|IP230-x
|extension module
|
|1000
|1000
|
|-
|IP241
|HW < 402 = 3
HW >= 402 = 2
|2400
|< 3500
|
|
|-
|IP222
|2
|2500
|4800***
|
|
|-
|IP232
|2
|2600
|4900***
|
|
|-
|IP111(A)
|1
|1400
|2000
|3800
|2500
|-
|IP112(A)
|2
|1800
|2100
|4000
|2500
|-
|IP101
|1
|
|
|2000
|
|-
|IP102
|2
|
|
|3500
|
|-
|}

*These values were averaged results from simulating normal and maximum use.
To have an official max total input power value please orientate on the upper limit of the associated PoE class.

**All devices leaving the manufacturing process are checked whether the cunsumption is under this value.

***LAN and PC interface connected with 1Gbit/s. 4xUSB connected (only IP2x2).

****LAN connected with 100Mbit/s. PC not connected. EEE active if supported. LCD light (idle state) == 0

Estimated worst case power consumption for operation of a 240-1000 with 3 IP230-X

5.9W +3*1W= 8.9W

Estimated worst case power consumption for operation of a 240 with 3 IP230-X

3.8W +3*1W= 6.8W

=== PoE Injector Capacity ===
All PoE injectors offered by innovaphone (order code 03-00010-228 and 03-00010-231) fullfill the power demand PoE capable innovaphone devices need (generally 15W at PSE side). This is good for any device up to PoE class 3.

==Supply via Power Supply Unit (PSU)==
Most of the innovaphone devices can be supplied by an external PSU (exception: IP800; IP2000; IP6000 with build in power supply).
All testreadings were determined by measuring the power consumption of the PSU. They are no official numbers. The following tables are showing different phone modes. All values in mW.
===12V PSU for phones===
Without using switch functionality (only LAN is connected 100Mbit/s):
{| border=1
|+
|Device
|Standby
|On-Hook
|Active (handsfree with max loudness)
|Ringing
|-
|IP110
|2900
|3000
|3400
|3000
|-
|IP200A
|3400
|3500
|3900
|3500
|-
|IP230
|3000
|3300
|3500
|3300
|-
|IP240
|3400
|3700
|3800
|3700
|-
|IP240-1000
|5200
|5500
|5700
|5500
|-
|IP241
|< 3500
|
|
|
|-
|IP222
|2500
|
|
|
|-
|IP232
|2600
|
|
|
|-
|}

Using switch functionality (PC + LAN is connected 100Mbit/s):
{| border=1
|+
|Device
|Standby
|On-Hook
|Active (handsfree with max loudness)
|Ringing
|-
|IP110
|3300
|3400
|3700
|3400
|-
|IP200A
|3900
|3900
|4400
|3900
|-
|IP230
|3400
|3600
|3800
|3600
|-
|IP240
|3800
|4100
|4200
|4100
|-
|IP240-1000
|7400
|7700
|7900
|7700
|-
|IP241
|4500***
|
|
|
|-
|IP222
|4200***
|
|
|
|-
|IP232
|4300***
|
|
|
|-
|}

====Use of extension modules====

* IP230/IP24x plus up to three IP230-X: a more efficient PSU is needed. Please use the 40V Gateway PSU or PoE injector from innovaphone (order no 03-00010-231)
* IP2x2 plus up to two IP2x2x: same as above - a more efficient PSU is needed. Please use the 40V Gateway PSU or PoE injector from innovaphone (order no 03-00010-231)
* IP2x2 plus one IP2x2x: POE Class 2 is sufficient if no other USB loads are present

===40V PSU for IP adapters and gateways===

Note as of May 2018: This product is End-of-life and no more offered by innovaphone.

The 40V PSU (order no. 03-00010-226) can be used for all gateways and adapters with small housing. In detail these are IP22, IP24, IP28, IP38, IP302 and IP305.

==Releated notes==
=== Heat output (BTU) ===
Sometimes we receive inquiries about the heat output (BTU - [https://en.wikipedia.org/wiki/British_thermal_unit British Thermal Unit]) of our devices.

innovaphone itself does not provide any information about this.

With afore mentioned power consumption values (! units are in mW) and corresponding converters (e.g. [https://umrechner.info/Kilowattstunde-BTU here]), the BTU can be calculated on your own responsibility.

Reference9:Event/0x00120004

2021-05-14T16:27:39Z

Cb: New page: This alarm indicates that the power supply does not provide enough power. There is e.g. only standard POE instead of POE plus available. Not all functions are available to allow operation ...

This alarm indicates that the power supply does not provide enough power. There is e.g. only standard POE instead of POE plus available. Not all functions are available to allow operation with lower power.

Reference9:Interfaces/PRI/Protocol

2020-03-04T09:36:14Z

Cb: /* QSIG D-channel protocol: */

In the '''Protocol''' submenu, you can set the protocol to be used for the ISDN interfaces. From the six available protocols, select the one that is best suited to your environment or if you have CAS enabled, select one of the three available:

== Euro ISDN D-channel protocol: ==
* '''EDSS1:''' This type of signalling has gained worldwide acceptance for ISDN subscribers and, despite the name, is also common outside Europe.

== QSIG D-channel protocol: ==
This is a standardised signalling method that is mainly used to connect PBXs. Here, basic call and tunnelling are supported by the gateways. This allows, in particular, homogeneous PBX systems to be linked with QSIG (1 byte), in which manufacturer-specific properties are exchanged via QSIG.
Unfortunately, there are several variants of the QSIG standard and various implementations; some conform more and some less to the standard.

The gateways therefore support two different variants.
* '''QSIG ECMA1:''' Numbering of the channels from 1-30 (2 bytes)
* '''QSIG ECMA2:''' Numbering of the channels from 1-15, 17-31 (2 bytes)

== NI D-channel protocol: ==
* '''NI:''' USA (National ISDN 1)
* '''5ESS:''' USA (AT&T)
* '''DMS100:''' USA (new standard)

== Mode: ==
If your trunk line is a point-to-point type, select Point-to-Point. If it is a point-to-multipoint connection, then select Point-to-Mulitpoint. This setting is irrelevant for permanent connections. If the connection is operated in mixed mode (one B channel permanently used for a fixed connection, one B channel in dial-up mode), the setting depends on the operating mode of the dial-up line (only TEL1-4 and PPP).
* '''Point-to-Point:''' Switches on the point-to-point connection.
* '''Point-to-Multipoint:''' Switches on the point-to-multipoint connection.

'''If CAS is selected under [[ Reference:Configuration/PRI/Physical | Configuration/PRI/Physical]], you have different options:'''

== CAS: ==
=== protocols: ===
* '''E1_R2D''': refers to the ITU-T standard Q.421-Q.422.
* '''E1_3BIT''': refers to an iranian standard, which is similar to E1_R2D
* '''T1_WINK_START''': refers to ANSI/TIA-464-C and works only with T1 (see [[ Reference:Configuration/PRI/Physical | Configuration/PRI/Physical]])
=== assign channels from top ===
channels will be assigned from top (first used channel 30 with E1 or 24 with T1).
=== channel bank mode ===
in the channelbank mode, each CAS channel is fixed assigned to a telephone number and is not dynamically assigned as in trunk scenarios. The CAS interface then is connected to a channelbank. E.g. telephone with number 1 calls to the CAS interface, channel 1 is taken for this call. If telephone with number 3 calls, channel 3 is taken and so on (this is normally done with maps in the routing table). In the incoming direction its similar. If a call comes in on channel 1, it is transferred to the telephone with number 1 (or via maps in the routing table).

Reference10:Interfaces/FXS/Physical

2020-02-12T10:51:06Z

Cb:

The '''physical''' settings of the analogue interfaces can be made here:

{|
|valign=top nowrap=true|'''Pulse:'''
| A checked check box enables the recognition of pulse dialling on the relevant interface.
|-
|valign=top nowrap=true|'''Reverse:'''
| A checked check box inverts the wiring of the relevant interface. This is only necessary in the event of incompatibility of the terminals, since some terminals (for example, in the US) are wired the opposite way.
|-
|valign=top nowrap=true|'''First ring followed by CLIP:'''
| This checkbox creates a short ring pulse of 160ms duration followed by a pause of 1 second. This pause is then used to send CLIP information to the phone. Then the ringing switches over to the normal intervals depending on the call being internal or external. If not checked the ringing directly starts with the corrosponding interval. Some telephones can explicitly suppress the first ring just to receive and present CLIP before audible ringing (often called ''1.st Ring OFF'' in menues). If a telephone cannot suppress the first ring or is not configured to do so, this checkbox may be unchecked to immediately start ringing without a short audible ring, which will occur otherwise. '''Telephones that do not accept CLIP information after the second ring need this checkbox checked to present CLIP correctly. One known model is the T-COM Concept P412.'''
|-
|valign=top nowrap=true|'''Receive gain:'''
| Set the loudness a user receives at this interface. Default is -10dB for short lines and -7 dB for long lines.
|-
|valign=top nowrap=true|'''Transmit gain:'''
| Set the loudness a user transmits at this interface. Default is -3dB for short lines and 0 dB for long lines. Some fax devices may need a lower transmit level to avoid clipping.
|-
|valign=top nowrap=true|'''Country:'''
| Set the line impedance for different countries. In Europe CTR21 is used.
|-
|valign=top nowrap=true|'''Internal/External call distinction:'''
|Normally internal and external calls are distinguished by terms of different rythms of the ringing voltage. Normally external calls use ''1sec ringing voltage - 4sec pause''. Internal calls normally use ''375ms ringing voltage - 250ms pause - 375ms ringing voltage - 4sec pause''. Select ''swiss'' to assign this rythms the opposite way. You can also select one rythm to be used permanently, no matter if the calls are external or internal.
|-
|valign=top nowrap=true|'''Metering pulse frequency:'''
|possible values : 16kHz or 12kHz. 16kHz is common for most countries. Austria and Swiss use 12kHz.
|-
|valign=top nowrap=true|'''Loop Current:'''
|As described before this topic changes the off-hook loop current and can be modified if 'Optimization grade' is set to 'low'. 'low' sets 20mA, 'mid' set 23mA, 'high' sets 26mA. Specific gateways may offer additional higher values.
|-
|valign=top nowrap=true|'''RingTrip Threshold:'''
|This topic modifies the current-threshold which lets the gateway distinguish between a ringing condition and a hook-off condition while the ringing voltage is applied from the gateway to the phone. When this threshold is exceeded by the phone, the gateway assumes a hook-off and therefore stops applying the ringing voltage and enters normal off-hook state. Phones that consume abnormal amounts of current while the AC ringing voltage is applied, cause the FXS to false detect a hook-off, which is quickly followed by a hook-on due to a missing loop current after the AC is switched off. This call is connected and released within fractions of a second and is lost. This typical symptom requires the ''RingTrip Threshold'' to be increased. 'low' sets 34mA, 'mid' sets 37,5mA and 'high' sets 40mA.
|}

====Receive and transmit sides====

{| border=4
|+
|FXS Gateway representing central office/PABX
|::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::direction:::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::::
|phone
|-
|PCM DAC
|>>>>>>>>>>>>>>>>>>>>>>>>>>>>>----receive direction---->>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>>
|phone speakerphone
|-
|PCM ADC
|<<<<<<<<<<<<<<<<<<<<<<<<<<<<<----transmit direction----<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
|phone microphone
|}

==== Some considerations about Flash, R and Dial-pulses ====

The a/b ports support further signaling through '''Flash'''-key (other sources call it '''R'''-key, what means exactly the same) followed by DTMF-Key combinations. For example, pressing '''Flash''' while already having a peer connection will set the current connection into call-hold state (the peer side hears ''Music-On-Hold'') and establish a new call. 

From a technical point of view '''Flash''' is a short interruption in offhook loop-current. You can imagine '''Flash''' as a very short hook-on/hook-off event. The on-hook duration of a '''Flash''' event is normally between 100ms and 300ms. Shorter durations are regarded as dial-pulses (if enabled), longer durations are regarded as seperate hook-on and hook-off events. If you encounter difficulties regarding '''Flash''', be sure you have configured a flash-time between 100ms and 300ms in your phone. 
 
'''Dial-pulse'''s are also short interruptions of the offhook loop-current and can also be imagined as short hook-on/hook-off events. The only difference to ''Flash'' is, '''Dial-pulse'''s have a much shorter duration of 40ms to 80ms. 
Several successive '''Dial-pulse'''s form a digit of a number. The pulses of one digit are seperated by a pause of about 60-80ms. Different digits are seperated with pauses that are much longer (at least 80ms, but can be much more). The digit '1' consists of one single pulse, '2' is two successive pulses and so on, '0' is ten successive pulses.

==== Pinout of the modular 6P2C a/b connector ====
[[Image:RJ11.gif]]

{| border=1
|+
|Pin
|Function
|-
|3
|TIP
|-
|4
|RING
|-
|}

==== Pinout of the modular 8P8C quad a/b connector ( IP29-20 )====

[[Image:RJ45.gif]]

Pinout and numbering follows TIA568B (like Ethernet)

{| border=1
|+
|Pin
|Function
|-
|1
| 2. PORT RING
|-
|2
| 2. PORT TIP
|-
|3
| 3. PORT RING
|-
|4
| 1. PORT TIP
|-
|5
| 1. PORT RING
|-
|6
| 3. PORT TIP
|-
|7
| 4. PORT RING
|-
|8
| 4. PORT TIP
|-
|}

Template:Leistungsmerkmale EN - Licenses Gateways

2018-11-03T15:37:05Z

Cb:

* IP29-x & IP38: includes appropriate amount of a/b and channel licenses
* IP311: includes 6 a/b and 6 channel licenses
* IP411: includes 2 a/b, 2 BRI and 6 channel licenses
* IP811: must be combined with up to 5 BRI (01-00500-002) and 10 channel licenses (01-00500-004)
* IP1130 & IP3011: must be combined with up to 1 PRI (01-00500-003) and 30 channel licenses (01-00500-004)
* IP6010: must be combined with up to 4 PRI (01-00500-003), 1 BRI (01-00500-002) and 60 channel licenses (01-00500-004)

Reference11r2:Event/0x001f0000

2018-09-20T16:29:14Z

Cb: Reference11r2:Event/0x001f0000 moved to Reference9:Event/0x001f0000

#REDIRECT [[Reference9:Event/0x001f0000]]

Reference9:Event/0x001f0000

2018-09-20T16:29:14Z

Cb: Reference11r2:Event/0x001f0000 moved to Reference9:Event/0x001f0000

'''DSP: Device status timeout'''

If the DSP doesnt send periodic status messages the DSP is restartet. This takes a fraction of a second, voice calls are not terminated.

If the DSP shows an error an event is generated, like:
06.10.2016-12:08:23 Error 0x001f0000 Indeterminate AC-DSP0 dsp status timeout

In addition a core dump of the DSP can be actived with the Web UI. Thick the checkmark
"DSP traceoff/coredump" at http://addr/debug.xml ->Tracing
Usually it is reasonable to tick also "DSP" and DSP control messages" and "DSP data messages"

If the core dump is written, another event is generated:
06.10.2016-12:08:32 Error 0x001f0000 Indeterminate AC-DSP0 core dump written, status=2

Then the core dump can be downloaded the http://addr/debug.xml -> Tracing -> download DSP core dump
The core dump is large, around 64Mbyte.
Once the coredump is downloaded the trace should be read and the system is ready to write another core dump if needed.

See also fix #14901 - IPxx11: Added DSP core dump option
Firmware 11r2 - 11r2 Service Release 20 (113635)
Firmware 12r1 - 12r1 Service Release 3 (120936)

Reference9:Event/0x001f0000

2018-09-20T16:00:31Z

Cb: New page: '''DSP: Device status timeout''' If the DSP doesnt send periodic status messages the DSP is restartet. This takes a fraction of a second, voice calls are not terminated. If the DSP sh...

Reference11r1:Interfaces/FXS/Signaling

2016-11-09T12:13:45Z

Cb:

The call '''signalling''' settings of the analogue FXS interfaces can be adjusted here:

{|
|valign=top nowrap=true|'''Disable:'''
| Disables the relevant analogue FXS interface.
|-
|valign=top nowrap=true|'''Speech Bearer Capability:'''
|Calls on the relevant interface are transmitted with Audio Bearer Capability as standard. A checked check box transmits calls from the relevant interface with Speech Bearer Capability. This only makes sense if only telephones are operated on the relevant interface (no fax machine or modem).
|-
|valign=top nowrap=true|'''Busy on Busy / No Call Waiting:'''
|A checked check box disables the call waiting signal for waiting calls on the relevant interface. Instead, Call busy is signalled to the calling side. This is necessary if, for example, a fax machine is operated on the relevant interface. ''Note that call waiting is generally only possible if 'feature codes' are enabled for this interface. It might be neccessary to switch on the 'call waiting' functionality explicitly with *43#. This checkmark refuses 'call waiting' even if activated within 'feature codes' and is a safe setting especially for fax machines.''
|-
|valign=top nowrap=true|'''Passive:'''
|Transfers the relevant interface to the passive mode. As a result, the Flash/Hook signal (R key) is not evaluated.
|-
|valign=top nowrap=true|'''No Call Transfer on Hook-On:'''
|A checked check box disables the call transfer function. Per default calls will be transfered on hook-on. Afterwards a call has been established, this call can be held and a new call can be initiated by pressing the R-Key. If the new call has been established or is yet ringing, the held and new call can be transferred together by hooking on the handset. There is a special exception : When the second call comes by ''call waiting'', what means, that you have your first call currently on hold, hooking on will NOT transfer the call waiting call with the previous, because this looks like an accident. If you really want to connect them you have to do this by 'R-4'. Hooking on will disconnect the call waiting call and the currently hold call will ring again (for 5 sec). If you use 'R-2' to switch between the call waiting call and the held call, this limitation is cleared and you may now transfer both calls together again by just hooking on.
|-
|valign=top nowrap=true|'''Signaling tones audibility:'''
|In normal operation in-band signaling tones are provided like well known from any phone provider. When the a/b is used in conjunction with a DOOR-Interface Converter, it may be desirable that the disconnect signaling tones at the end of a call are not audible to the person at the door. Checking 'in call establishing phase only' does so, but does not suppress the signalling tones in call establishing phase. This implies that the person at the door will hear typical telephone sounds after pressing the doorbell-button. This mode additionally disables recognition of flash (R) pulses. Checking 'never/silent' additionally supresses singalling tones in call establishing phase, so that there will be no audible telephone sounds at all to the person at the door. This mode disables recognition of flash (R) AND DTMFs as well. This might be the typical choice for DOOR-Interfaces.
|-
|valign=top nowrap=true|'''Reverse polarity in conversation/connected state:'''
|If checked, the gateway will reverse the line polarity to 'conversation/connected state polarity' as soon as the connection to the peer side is established. As a result of this, the line polarity returns to normal polarity again if the peer hooks up the connection. In addition a busy tone is provided. Equipment connected to the FXS port may take advantage of this additional information and safely detect call state details.
|-
|valign=top nowrap=true|'''Volume:'''
|Sets the output volume to the relevant interface (from IP to FXS), in decibel (dB), between -32dB and +32dB. No value or the value 0 is equal to the factory settings.
|-
|valign=top nowrap=true|'''Metering pulse every:'''
|The gateway can generate periodic metering pulses in fixed intervals of time. The value specified here is the time between two successive pulses. A value of 0 or no value disables pulse generation. ''Note that metering pulses are only generated for outgoing calls. The first metering pulse is generated when the call is connected. Note also that enabling pulse metering implicitly switches on '''Passive''' and '''No Call Waiting''', so that no call can be held and no incoming caller is accepted within a call.''
|-
|valign=top nowrap=true|'''Caller ID 1 standard:'''
|CallerId/CLIP can be sent in different ways depending on the country. The basic difference is the type of modulation used. Possibilities are DTMF (Dual Tone Multiple Frequency) and FSK (Frequency Shift Keying). FSK modulation splits up into the Telcordia/Bellcore and ETSI scheme. DTMF modulation has several subsidaries and differ in special digits preceding and succeeding the number information.
 Examples using 1234567890 as calling number :
 ETSI DTMF : A1234567890C or D1234567890C
 Danish DTMF : A1234567890* or D1234567890*
 Indian DTMF : 1234567890
 Brazilian DTMF : A11234567890C or AB1234567890C
 The gateway will add these additional digits to the calling number as shown above, depending on the standard selected.
|}

Reference11r1:Interfaces/FXO/Signaling

2016-11-09T12:13:05Z

Cb:

The call '''signalling''' settings of the analogue FXO interfaces can be made here:

{|
|valign=top nowrap=true|'''Disable:'''
| Disables the relevant analogue FXO interface.
|-
|valign=top nowrap=true|'''Speech Bearer Capability:'''
|Calls on the relevant interface are transmitted with Audio Bearer Capability as standard. A checked check box transmits calls from the relevant interface with Speech Bearer Capability. This only makes sense if only telephones are operated on the relevant interface (no fax machine or modem).
|-
|valign=top nowrap=true|'''Central Office Tone detection:'''
|When the FXO initiates a call, normally a audible tone is expected to be received from the central office/local PABX side. The tone detection works independantly from any country setting. When the expected tone cannot be detected within 2.6sec, then the call will be cancelled. If this tone detection is disabled, the FXO will continue dialing DTMF tones of the configured number, with a fixed delay of 800ms after hook-off.
|-
|valign=top nowrap=true|'''Terminate ringing calls which failed to reach its destination by offhook-onhook sequence:'''
|Calls that are ringing on the FXO port and for any kind of reason failed to reach its destination on the IP side normally have no chance to get informed of that miss. Instead the call would stay in ringing state on the FXO port for long time. This option allows the FXO to react on a failed call to IP with a short offhook-onhook sequence on the FXO port and therefore terminate the ringing call immediately.
|-
|valign=top nowrap=true|'''Establish call on line polarity reversal:'''
|When the FXO port detects a reversed polarity and the FXO port is idle, a new call wil be established. Countries that use '''DTMF CallerId''' also often use a '''Polarity Reversal''' preceding the DTMF CLIP sequence and start with AC ringing afterwards. In detail the sequence is as follows : Line polarity reversed, 200ms pause, DTMF CallerId transmission, Line polarity normal, 1000ms pause, AC ringing, pause ... . For instance Sweden is one of those countries that use DTMF CallerId combined with Line polarity reversal call establishment. If this checkmark is unchecked here, the call will miss the DTMF CallerId sequence, when established with the AC ringing. Note that this option can also be used with FSK CallerId.
|-
|valign=top nowrap=true|'''PSTN drives conversation state in reversed line polaritity:'''
|Calls from FXO->PSTN normally switch to conversation/connected state within 4seconds after dialing. If this option is checked AND the PSTN changes its line polarity while a connection to the dialed peer is established, then the FXO can recognize the reversed line polarity and switch to connected state prior to the elapse of 4seconds. Once in connected state, the FXO can detect line polarity switched back to normal polarity and assume a peer disconnect without having detected any kind of call-progress-tone (e.g. busy-tone). This is faster and in some countries the busy-tone at the end of a call is simply omitted, when polarity switching is used. If you use '''Establish call on line polarity reversal''', be shure if your provider additionally uses polarity reversals in conversation/connected state or not. If your provider does, and you miss to check this option, this might lead to strange but short and self-ending re-calls after each hook-on. This comes from the PSTN's final switch to the normal line polarity, which then leads to a wrong call. When checked, the FXO will wait for the PSTN's line polarity change at the end of the call. This behaviour is common for PSTN in Sweden.
|-
|valign=top nowrap=true|'''Volume:'''
|Sets the output volume to the relevant interface (from IP to FXO), in decibel (dB), between -32dB and +32dB. No value or the value 0 is equal to the factory settings.
|-
|valign=top nowrap=true|'''Caller ID 1 standard:'''
|Selects the standard in which '''CallerId/CLIP''' is detected and decoded. '''CallerId''' carries the caller's phone number in case of an incoming/ringing call. This number is fed into the resulting voip call as the CGPN (Calling Party Number). Note that for this reason incoming calls from the FXO port must use ''en-bloc'' delay dialling to allow for receipt of the CLI. This may be done by registering the Port with a ''Gateway'' type PBX object and set the ''Enblock Count'' option in this object or use a route with ''Force enblock'' set in the ''Gateway->Routes'' Menu.

'''FSK CallerId''' standards can contain additional information like ''Called Line Id'' or ''Date/Time'' information. These additional informations will be dropped. '''DTMF CallerId''' only contains the ''Calling Line Id''.
|}

Reference10:General/Info

2016-06-07T11:57:44Z

Cb:

General information about the VoIP device is displayed here.

== All Devices ==

{|
|valign=top nowrap=true|'''Version'''
|The version string of the device. This string has the format <Version><label><Product>[<Firmware Build>], Bootcode[<Bootcode Build>], HW[<Hardware Build>], <Fash>/<Dram>
|-
|valign=top nowrap=true|'''SerialNo'''
|The serial number or Ethernet MAC address of the device. It consists of 6 pairs of hexadecimal digits separated by '-'. It is followed by a two digits checksum, which can entered on the license manager to detect possible typing errors when entering the serial no.
|-
|valign=top nowrap=true|'''Coder'''
|Voice Coder channels available. Note that these figures show type and number of channels the device technically can handle. It does not account for possible license requirements (that is: if a license for a coder is required but not available, the coder would show up here nevertheless)
|-
|valign=top nowrap=true|'''HDLC'''
|Number of HDLC channels available.
|-
|valign=top nowrap=true|'''Sync'''
|Some devices can synchronize the internal PCM bus (the bus used to interconnect ISDN channels, DSP channels, etc) to an external interface. If no synchronization is achieved, this may result in slips on ISDN interfaces, which results in a removal or duplication of a sample every some seconds. In voice calls this is not recognized, but it may lead to abortion of Fax or Modem calls.
|-
|valign=top nowrap=true|'''Power Source*'''
|Indicates which ETH Interface it's being used for PoE power source. Note*: Some Gateways don't provide this information.
|-
|valign=top nowrap=true|'''Power Off Loop*'''
|Provides information if feature it's Enabled/Disabled. Note*: Only some devices have this feature and provide information.
|-
|valign=top nowrap=true|'''Temperature'''
|Displays the temperature of the mainboard inside the Gateway. Up to 60ºC it's an acceptable value, higher values than that we recommend to move the Gateway to a cooler location to prevent damages on the hardware. IP111 IP112 and IPxx11 devices show the CPU die temperature, it can rise to 85°C.
|-
|valign=top nowrap=true|'''SNTP Server'''
|The SNTP Server used to synchronized time/date of the device
|-
|valign=top nowrap=true|'''Time'''
|Current Date/time as synchronized with an external SNTP Server
|-
|valign=top nowrap=true|'''Uptime'''
|Uptime of the device
|-
|valign=top nowrap=true|'''Test License'''
|Link, which is displayed if a test license is installed on the device. This link can be used to restart the test period.
|}

== DECT Devices ==

{|
|valign=top nowrap=true|'''Firmware'''
|The Firmware Version of the DECT subsystem.
|-
|valign=top nowrap=true|'''System ARI'''
|The ARI of the DECT system. On single cell systems this is a burned in identification of the system (similar to a MAC address). For multicell systems the ARI is defined by the multicell license, which has to be installed on the ip-master of the system.
|-
|valign=top nowrap=true|'''Frequency'''
|The frequency used on the system. This is DECT for systems conforming to the DECT standard. For north america a different frequency is used.
|}

ReleaseNotes:Hardware IP232

2016-04-11T21:26:51Z

Cb: New page: This roadmap describes IP2x2x hardware build starting at build 400 This article is generated automatically. Do not edit! Please see ''[[Support:What is the DVL-Roadmap?|the dis...

This roadmap describes IP2x2x hardware build starting at build 400 
 

This article is generated automatically. Do not edit!
Please see ''[[Support:What is the DVL-Roadmap?|the disclaimer]]'' before using the information presented here!

__NOEDITSECTION__
{{#invoke-url: http://wiki.innovaphone.com/extensions/InvokeUrlFunction/generateProjectList.php?area=fixes&category=Hardware%20IP232}}

2014-12-10T08:45:28Z

Cb:

==Summary==

A dedicated headset interface isn’t available on the IP111 but it is still possible to connect a headset.

==Applies To==
This information applies to
*IP111

==More Information==
===Problem Details===

The headset is connected to the IP111 handset port.
The IP111 can be configured to headset mode, then the headset key can be used a hook switch and the handset equalizer is disabled.

=== Pinout and signal definition ===

PIN Assignment of the IP111 handset modular 4P4C jack is :

[[Image:RJ9.gif]]
{| border=1
| 1 || Mic +
|-
| 2 || Receiver
|-
| 3 || Receiver
|-
| 4 || Mic -
|}

The microphone is driven symetric with 2.9V 1.35kOhm. Microphone sensivity should be -42db (V/Pa)
The Receiver impedance should be 150Ohm typ, 32Ohm min.

=== Headsets ===

Since the microphone is connected to a symetric amplifier it is important to use a headset with symetric cabling.

This headsets should work, the marked headset has been tested without problems:
:BIZ-2300 Mono Balanced 2303-825-109
:BIZ-2300 Duo Balanced 2309-825-109 (*)
:BIZ-2400 Mono Balanced 2486-825-109
:BIZ-2300 Duo Balanced 24869-825-109

The cable should be 8800-00-25A, with modular 4P4C connector on one side and headset connector on the other side.

[[Category:Howto|{{PAGENAME}}]]

2012-05-29T13:43:36Z

Cb:

Reference9:Interfaces/ETH/Link

2012-05-29T13:39:58Z

Cb: /* Configuration */

Howto:PIN Assigment for IP200 Headset connector

2012-04-02T14:01:13Z

Cb: /* Configuration */

==Applies To==
This information applies to

* IP200
* IP200A



==More Information==

=== Configuration===
PIN Assignment of the IP200 headset connector

<pre>
_||_
|1234|
||||||
</pre>

Front view, flag on the top.

{| border=1
| 1 || Mic -
|-
| 2 || Speaker
|-
| 3 || Mic +
|-
| 4 || Speaker
|}

The microphone is driven symetric with 1.6V 2kOhm.
The Speaker port impedance is 100Ohm.



[[Category:Howto|{{PAGENAME}}]]

Howto:PIN Assigment for IP200 Headset connector

2012-04-02T14:00:51Z

Cb: /* Configuration */

==Applies To==
This information applies to

* IP200
* IP200A



==More Information==

=== Configuration===
PIN Assignment of the IP200 headset connector

<pre>
_||_
|1234|
||||||
</pre>

Front view, flag on the top.

{| border=1
| 1 || Mic -
|-
| 2 || Speaker
|-
| 3 || Mic +
|-
| 4 || Speaker
|}

The microphone is driven symetric with 1.6V 2KOhm.
The Speaker port impedance is 100Ohm.



[[Category:Howto|{{PAGENAME}}]]

Reference9:Interfaces/PRI/Physical

2012-03-09T11:07:45Z

Cb: T1 Tx pulse shape description changed

The basic settings of the '''ISDN Primary Rate interfaces''' ('''PRI''') can be made here. The visibility of some settings depends on other settings.

* '''NT Mode:''' The interface is operated in TE (Terminal Equipment) mode as standard. It behaves like a normal ISDN terminal and synchronises itself to the network clock (clock slave). A checked check box operates the interface in NT (Network Termination) mode. It behaves like an ISDN network termination (NTBA) and provides the clock pulse (clock master).
* '''Clock Mode :'''
<blockquote>
* Derived from NT mode (Master/Slave configuration depends on setting of '''''NT Mode''''' checkbox : NT Mode => Master, otherwise Slave).
* Slave (the device obtains its clock pulse from its peer device, a clock master).
* Master (the device provides the clock pulse).
</blockquote>
* '''Swap tx/rx:''' Switches the interface of the transmit line with that of the receive line.
* '''Do not use for synchronisation:''' Slave Mode interfaces are by default used to obtain the clock for the internal PCM bus. With this checkmark the interface can be excluded as clock source. This can be checked to control where the device gets its clock from. (For a direct cable from one PRI interface to another PRI interface of the same box, this checkmark has to be set on the slave interface for such a connection, because synchronization does not work).
* '''µ-law:''' This check box must be checked if the device is located in a country that uses the ISDN µ-law standard. This includes the North America and Asian region, for example.
* '''T1 :''' Switches the interface from the E1 European standard (2.048MBit/s or 30 ISDN B channels) to the T1 American standard (1.544 MBit/s or 24 ISDN B channels), which is used in the US, Canada and Japan.
* '''CAS :''' Use CAS signaling instead of ISDN signaling. With the CAS method (Channel Associated Signalling), the signalling data is transferred in 4 bits (ABCD) instead of messages. On E1 lines CAS is signaled on channel 16 (D channel). On T1 lines the CAS information is send on the individual B-channel in the LSB every 6th frame. T1 in CAS mode has 24 voice channel ( instead of 23 voice channels with ISDN ).

'''Caution'''
If you enable the CAS method, the content of chapter
"[[Reference:Configuration/PRI/Protocol|Configuration/PRI1-n/Protocol]]" changes.

* '''No CRC4 :''' A checked check box disables the Cyclical Redundancy Check. Select ''No CRC4'' only in the rare cases the links supports no CRC.
* '''Activate 'power-off' relay :''' Closes the relay of the power-off loop. This performs a physical connection of the RJ45 jack with the jack of the next interface. Checking this check box also puts the two corrosponding PRI interfaces into high impedance state, so that they appear as disconnected from their RJ45 jacks.
* '''Loopback :''' A checked check box enables the loopback function. This is only necessary for test purposes. The loopback function simulates a connected device on the relevant interface.
* '''Tx attenutation for T1 mode :''' Allows the transmit signal level for T1 mode to be decreased for short lines. '0db' selects the TIA968 option A output pulse, this is the "normal" pulse with full amplitude. '7.5db' and '15db' selects the attenuated TIA 968 Option B and C output pulses and is used if a legacy repeater is connected with a short cable. Only visible if T1 is checked. For details ask the operator or read the TIA968 or "O'Reilly T1. A Survival Guide" chapter6.
* '''Send flags on FDL :''' A checked check box transmits FDL messages (Facility Data Link). Only visible if T1 is checked. The T1 link managment protocol AT&T TR 54016 ( also called Telemetry Asynchronous Block Serial Protocol or TABS ) needs flags on the FDL to work reliable.

==== Pinout of the PRI connector ====
RX and TX can be swapped for NT operation with software configuration so that no crossover cable is needed.
Transmit and receive signal are for 120Ohm symetric interfaces.
See also ISO/IEC 10173 and USOC RJ48-C.

TE Mode or NT Mode with 'Swap tx/rx' checked

{| border=1
|+
|Pin
|Function
|-
|1
|RX+
|-
|2
|RX-
|-
|4
|TX+
|-
|5
|TX-
|-
|}
[[Image:RJ45.gif]]

NT Mode or TE Mode with 'Swap tx/rx' checked

{| border=1
|+
|Pin
|Function
|-
|1
|TX+
|-
|2
|TX-
|-
|4
|RX+
|-
|5
|RX-
|-
|}
[[Image:RJ45.gif]]

Reference:Configuration/ETH/Link

2011-10-31T21:40:18Z

Cb: /* Configuration */

Howto:T.38 Redundancy

2011-03-14T10:32:18Z

Cb: /* Problem Details */

==Applies To==
This information applies to

* IP 2x, IP30x, IPxx10

Firmware V8 hotfix 12 and later



==More Information==

===Problem Details===
Jitter and packet loss can ruin T.38 fax transfer. If bandwidth is not an issue but packet loss is, T.38 redundancy can help. The idea is to re-transmit certain packets or to proactively transmit certain packets several times in order to increase the reliability of the T.38 transmission.

There are 3 mechanism implemented:

; Retransmission : when a signal ( e.g. CED or V21-preamble) is transmitted and it is retransmitted after a certain time (''T38 noop-period'').
; Slow Speed Data Duplication : slow speed (V.21) control data is sent 2 or 3 times (hoping one of the duplicates will make it to the destination in time). With each message a copy of the previous or the 2 previous payloads are sent. So the data rate is increased, but not the packet rate.
; High Speed Data Duplication : high speed image data is sent 2 or 3 times (hoping one of the duplicates will make it to the destination in time)

===Configuration===
T.38 redundancy parameters can be configured using the interface shown at <code>http://x.x.x.x/AC-DSP0/mod_cmd.xml?xsl=dsp.xsl</code>.

===Known Problems===

Data duplication may increase the likelihood that all relevant packets make it to the destination. However, it increases the bandwidth actually used and therefore may in fact have a negative effect on reliability itself, making things worse than before. Especially high speed data duplication may be harmful.

Also, fax G3 devices feature ECM (error correction mode). This will retransmit image data if corruption is detected. High speed data duplication may be obsolete thus.

Finally, some 3rd party fax devices get confused by duplicate or retransmitted T.38 data. As a result of that, T.38 redundancy must be set carefully according to the individual site situation.



[[Category:Howto|{{PAGENAME}}]]