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VoIP EP command reference
VoIP phone ports
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iMG/RG Software Reference Manual (Voice Service)
A customized ring signal with a complext cadence. The ring is executed three times with
a cadence of 0.5 sec. on and 0.5 sec. off followed by an infinite cadence of 1 sec on and 2
sec off. The timeout is set to 180 sec.
--> voip ep signaling create create myring type ring time-out 180 fre-
quency 25 cadence -3(+0.5-0.5)+continuous(+1.0-2.0)
See also
voip ep SIGNALING ADD
voip ep SIGNALING DELETE
voip ep SIGNALING LIST
voip ep SIGNALING REMOVE
voip ep SIGNALING SHOW
6.3.3.1.50 VOIP EP SIGNALING DELETE
Syntax
VOIP EP SIGNALING DELETE <name>
Description
This command deletes an entry in the customized signaling list.
Options
The following table gives the range of values for each option, which can be specified with
this command, and a default value (if applicable).
Example
--> voip ep signaling create delete dial1
See also
voip ep SIGNALING ADD
voip ep SIGNALING CREATE
voip ep SIGNALING LIST
voip ep SIGNALING REMOVE
voip ep SIGNALING SHOW
6.3.3.1.51 VOIP EP SIGNALING LIST
Syntax
VOIP EP SIGNALING LIST
Description
This command lists all the entries in the customized signaling list defined in the system
using the VOIP EP SIGNALING CREATE command.
The following information is displayed:
Option
Description
Default Value
name
A name that identifies an existing customized signal-
ling created with the VOIP EP SIGNALING CREATE
command. To display the existing access port names,
use the voip ep list command.
N/A
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VoIP phone ports
VoIP EP command reference
iMG/RG Software Reference Manual (Voice Service)
6-118
signaling entry ID value
signaling entry name
signaling entry type
Example
--> voip ep signaling list
Custom Signaling Protocol items:
ID
|
Name
|
Type
-----|------------|---------------------------------------
1 | mydial
| dial-tone
2 | mybusy
| busy-tone
3 | myring
| cai
-----|------------|---------------------------------------
See also
VOIP EP SIGNALING ADD
VOIP EP SIGNALING CREATE
VOIP EP SIGNAaLING DELETE
VOIP EP SIGNAaLING REMOVE
VOIP EP SIGNALING SHOW
6.3.3.1.52 VOIP EP SIGNALING REMOVE
Syntax
VOIP EP SIGNALING REMOVE <name> PORT <port>
Description
This command removes a previously added customized signaling from an existing end-
point.
Options
The following table gives the range of values for each option, which can be specified with
this command, and a default value (if applicable).
Example
--> voip ep signaling remove myring port prt0
See also
voip ep SIGNALING ADD
voip ep SIGNALING CREATE
Option
Description
Default Value
name
A name that identifies an existing customized sig-
nalling created with the VOIP EP SIGNALING
CREATE command. To display the existing access
port names, use the voip ep list command.
N/A
port
A name that identifies an existing access port. To
display the existing customized signalling use the
voip ep SINALING list command.
N/A
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VoIP EP command reference
VoIP phone ports
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iMG/RG Software Reference Manual (Voice Service)
voip ep SIGNALING DELETE
voip ep SIGNALING LIST
voip ep SIGNALING SHOW
6.3.3.1.53 VOIP EP SIGNALING SHOW
Syntax
VOIP EP SIGNALING SHOW <name>
Description
This command shows a previously created customized signaling.
The following information is displayed:
signaling entry type
signaling entry time out
signaling entry frequency
signaling entry cadence
signaling entry attached endpoints
Options
The following table gives the range of values for each option, which can be specified with
this command, and a default value (if applicable).
Example
--> voip ep signaling show mydial
Custom Signaling Protocol item: mydial
----------------------------------------------------------
Type:
dial-tone
Time-Out:
Frequency:
240x340/425x525/340+480 Hz.
Cadence:
-3(+0.5-0.5)+continuous(+1-1)
Attached ports:
tel1
See also
voip ep SIGNALING ADD
voip ep SIGNALING CREATE
voip ep SIGNALING DELETE
Option
Description
Default Value
name
A name that identifies an existing customized signal-
ling created with the VOIP EP SIGNALING CREATE
command. To display the existing access port names,
use the voip ep list command.
N/A
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Common VoIP attributes: QoS, Media and DTMF-Relay
QoS
iMG/RG Software Reference Manual (Voice Service)
6-120
voip ep SIGNALING LIST
voip ep SIGNALING REMOVE
6.4
Common VoIP attributes: QoS, Media and DTMF-Relay
The following section details voip attributes that are typically used to handle voice quality and that are common
for both the two main VoIP protocols (SIP and MGCP).
In some specific cases some commands are effective only when a specific voip protocol is selected
(see the
comand details).
All the following commands can be entered only after the voip protocol has been enabled.
6.4.1
QoS
iMG/RG/iBG devices allow to specify a specific DSCP/TOS priority value to the originated VoIP traffic.
DSCP and TOS are mutually exclusive because they refers to the same IP Header field using only a different
number of bits (3 bits in case of TOS, 6 bits in case of DSCP) and assigning different packet classification accord-
ingly to the TOS or DSCP value.
The command VOIP QOS SET DSCP is used to set the DSCP value while the VOIP QOS SET TOS command
is used to set the TOS value. The two commands are mutually exclusive: only DSCP or TOS can be configured.
Attempting to configure the TOS attributes after having configured the DSCP attributes (or viceversa) will
result in a conflict failure. It’s necessary reset the DSCP attributes (or TOS) befor setting the TOS attributes
(or DSCP).
It’s possible differentiate DSCP/TOS values for VoIP signalling packets from the
DSCP/TOS values for the media
streams.
The command voip qos set dscp/tos signaling-protocol specify the DSCP/TOS values only for VoIP signaling
packets originated by iMG/RG/iBG devices.
The command voip qos set dscp/tos media-protocol specify the DSCP/TOS values only for VoIP media packets
originated by iMG/RG/iBG devices.
If signalling-protocol or media-protocol is not specified, the DSCP/TOS values are set to the same value for
both signaling and media packets.
6.4.2
Media
iMG/RG/iBG devices allow to specify a specific pool of IP UDP ports to be used for media (RTP) transport.
Each even media port is paired with the subsequent odd portand assigned to RTP or RTCP streams respec-
tively.
Then maximum number of simultaneously streams is therefore half of the size of the media range.
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DTMF-RELAY
Common VoIP attributes: QoS, Media and DTMF-Relay
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iMG/RG Software Reference Manual (Voice Service)
When VoIP
MGCP protocol is used, the lowest ports pair is assigned to the first configured end-point, the sub-
sequent pair is assigned to the second configured end-point and so on.
When SIP protocol is used, ports pair are used in a round robin fashion.
iMG/RG/iBG devices allow to specify a specific the RTP packetization time as timeframe between to consecutive
RTP packets.
Packetization time could be negotiated at runtime during the call establishment phase.
The value specified via CLI is the value that is normally advertized by the VoIP protocol (via VoIP signalling mes-
sages) and that it’s used when is not negotiated during the call setup.
6.4.2.1 Media Timeout
Only when VoIP SIPprotocol is used, it's also possible set iMG/RG/iBG devices to detect if an incoming RTP flow
is still present or not (e.g. the other end-point was abruptly disconnected or network has critical problems)
forcing the call release if no RTP packet flow has been detected for the current call for a time longer than the
specified observation period.
6.4.3
DTMF-RELAY
iMG/RG/iBG devices support DTMF relay for the transmission of DTMF dialpad tones.
DTMF tones can be transmitted in-band through the RTP media stream as normal voice or can be transmitted
via RTP stream accordingly to RFC2833 or can even being transmitted via VoIP specific messages.
Using DTMF in-band transmission is reliable only when the media stream uses uncompressed codec like G.711u
or G.711a. When compressed codecs
like G.729 or G.726 are used, the distortion produced by the codec
compression algorithm does not allow a clear transmission of the DTMF tones that could be not intelligible on
the far end.
In this case iMG/RG/iBG devices allow to use RFC2833 Telephone events or SIP NOTIFY or MGCP NTFY mes-
sages for DTMF transmission.
In case of SIP protocol, it’s possible force the system to automatically select the proper DTMF relay mode
depending on the negotiated codec.
In case of MGCP protocol, because call properties are controlled by the call agent, the selection of the proper
DTMF relay mode is demanded to the call agent logic.
6.4.4
Functional Differences for Common VoIP attributes in Product Categories
The table below is intended to identify what is common amongst the product families - as well as where there
are differences - to highlight those differences. To determine which family your device belongs to - please refer
to the preface.

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