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VoIP SIP command reference
VoIP SIP
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iMG/RG Software Reference Manual (Voice Service)
Users previously registered to location servers start to unregister themselves and then
re-register on the same location servers.
This command doesn't release any resources (users, physical ports and FDB entries) pre-
viously created during module configuration.
Example
--> voip sip protocol restart
See also
VOIP SIP PROTOCOL ENABLE
6.2.4.1.4 VOIP SIP PROTOCOL SET AUTHENTICATION
Syntax
VOIP SIP PROTOCOL SET AUTHENTICATION {PROXY | PROXY,WWW | WWW}
Description
This command sets the SIP dialog authentication method. By default, this is set to PROXY.
Example
--> voip sip protocol set authentication proxy
See also
VOIP SIP PROTOCOL SHOW
VOIP SIP PROTOCOL ENABLE
6.2.4.1.5 VOIP SIP PROTOCOL SET CONTACT-ON-1XX-RESPONSE
Syntax
VOIP SIP PROTOCOL SET CONTACT-ON-1XX-RESPONSE {ENABLE | DIS-
ABLE}
Description
This command sets enables and disables the inclusion of a Contact header in SIP 1xx
responses. By default, this is set to disabled.
Example
--> voip sip protocol set contact-on-1xx-response enable
See also
VOIP SIP PROTOCOL SHOW
6.2.4.1.6 VOIP SIP PROTOCOL SET DEFAULTPORT
Syntax
VOIP SIP PROTOCOL SET DEFAULTPORT <ipport>
Description
This command sets the default listening/sending port used for SIP
signalling
messages.
By default, when the SIP module is attached to an IP interface using theVOIP SIP PROTO-
COL SET NETINTERFACE command, the following default value is used:
defaultport:5060
Note:
Changing the signalling port causes the SIP module to restart.
Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
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Example
--> voip sip protocol set defaultport 5060
See also
VOIP SIP PROTOCOL ENABLE
6.2.4.1.7 VOIP SIP PROTOCOL SET EXTENSION
Syntax
VOIP SIP PROTOCOL SET EXTENSION <extension>
Description
This command sets extended protocol features.
Note:
100rel and Session Timer are always supported when requested; setting “session-timer” the user agent
explicitly requires this keep-alive mechanism. Info method overlaps the event transfer supported by RTP
sessions.
Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
Example
--> voip sip protocol set extension session-timer
See also
VOIP SIP PROTOCOL SHOW
6.2.4.1.8 VOIP SIP PROTOCOL SET INFO
Syntax
VOIP SIP PROTOCOL SET INFO {DTMF_0 | DTMF_1 | DTMF_2 | DTMF_3 |
DTMF_4 | DTMF_5 | DTMF_6 | DTMF_7 | DTMF_8 | DTMF_9 | DTMF_Star |
DTMF_Gate | DTMF_A | DTMF_B | DTMF_C | DTMF_D | Flash} <token>
Description
This command sets mappings for out of band DTMF digits and flash-hook signals within
SIP INFO methods using the application/dtmf-relay content type.
Option
Description
Default Value
ipport
UDP/TCP port number used for signalling mes-
sages. Available values are in the range 1026 to
65534.Only even values can be accepted
5060
Option
Description
Default Value
extension
Extension is comma separated list of values
defining the protocol extension. Available values
are:
info
session-timer
keep-18x-session
none
none
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Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
Example
--> voip sip protocol set info DTMF_0 Signal=0
See also
VOIP SIP PROTOCOL SHOW
6.2.4.1.9 VOIP SIP PROTOCOL SET INTERNAL-CALL-ROUTING
Syntax
VOIP SIP PROTOCOL SET INTERNAL-CALL-ROUTING {ENABLE | DISABLE}
Description
This command enables/disables the internal-call-routing feature. By default, if more than
one SIP user has been created on the iMG device, a call between two of them does not
contact the configured SIP proxy server. By setting internal-call-routing to disable, the
device always contacts the SIP proxy server.
Example
--> voip sip protocol set internal-call-routing disable
See also
VOIP SIP PROTOCOL ENABLE
6.2.4.1.10 VOIP SIP PROTOCOL SET INVITETIMEOUT
Syntax
VOIP SIP PROTOCOL SET INVITETIMEOUT <sec>
Description
This command sets the number of seconds an INVITE that does not receive any answer
must be sent. During an outgoing call, the INVITE sent by iMG must be received an
answer within ROUNDTRIPTIME msec. If the answer is not received, the same INVITE is
re-transmitted and the ROUNDTRIPTIME’s value is doubled. This process ends INVITE-
TIMEOUT seconds after the first INVITE.
Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
Option
Description
Default Value
token
Signal element to be used for the DTMF
digit of flash-hook event. These should take
the form ‘Signal=
x
’, where
x
is one of the
digits 0-9, *, #, A, B, C, or D.
N/A
Option
Description
Default Value
sec
Number of seconds that an INVITE with no
answer must be sent.
32
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Example
--> voip sip protocol set invitetimeout 10
See also
VOIP SIP PROTOCOL SET ROUNDTRIPTIME
6.2.4.1.11 VOIP SIP PROTOCOL SET KEEP-ALIVE
Syntax
VOIP SIP PROTOCOL SET KEEP-ALIVE {ENABLE|DISABLE}
VOIP SIP PROTOCOL SET KEEP-ALIVE TIME <sec>
Description
This command sets a keep-alive mechanism based on the REGISTER message. When the
feature is enabled, iMG sends a REGISTER message every KEEP-ALIVE TIME seconds.
Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
Example
-> voip sip protocol set keep-alive 150
See also
VOIP SIP PROTOCOL SET ENABLE
6.2.4.1.12 VOIP SIP PROTOCOL SET NAT
Syntax
VOIP SIP PROTOCOL SET NAT ADDRESS {<host> | NONE}
VOIP SIP PROTOCOL SET NAT INTERFACE <interface>
Description
This command sets the NAT host reference. Any SIP URLs with local references are hid-
den by the NAT address value. It also specifies the external interface for NAT to use.
Note:
Changing the NAT reference causes the SIP module to restart.
Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
Option
Description
Default Value
sec
Interval between two REGISTER messages.
300
Option
Description
Default Value
host
The address that will be displayed in the local SIP URL ref-
erences. It can be expressed in hostname format or IPv4
format. A Hostname can be at most 255 characters long.
None
interface
The iMG interface that NAT is to use as the external
interface.
None
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Example
--> voip sip protocol set nat address iMG.voip.atkk.com
--> voip sip protocol set nat interface ip0
See also
IP LIST INTERFACES
VOIP SIP PROTOCOL ENABLE
VOIP SIP PROTOCOL SHOW
6.2.4.1.13 VOIP SIP PROTOCOL SET NETINTERFACE
Syntax
VOIP SIP PROTOCOL SET NETINTERFACE <interface_name>
Description
This command sets the IP interface used to access the VoIP network.
Signalling
and voice packets will use the Source IP address defined for the selected inter-
face.
Options
The following table gives the range of values for each option that can be specified with
this command and a default value (if applicable).
Example
--> voip sip protocol set netinterface ip0
See also
VOIP SIP PROTOCOL ENABLE
6.2.4.1.14 VOIP SIP PROTOCOL SET PATH-HEADER
Syntax
VOIP SIP PROTOCOL SET PATH-HEADER {ENABLE|DISABLE|PROFILE}
Description
This command includes the PATH-HEADER support on REGISTER messages as detailed
by RFC 3327 to discovering intermediate proxies during SIP registration.
By default the path-header support is tied to the SIP profile defined at admin level. If the
SIP profile requests the path-header support, then the support is automatically turned on,
otherwise it is left off.
It’s also possible to force the path-header to be always turned off or on independently
from the profile selected.
When path-header support is enabled and the iMG receives a valid path-header value a
response to the REGISTER request, all the subsequent outgoing calls will use the address
specified by the path-header value as outgoing proxy.
Option
Description
Default Value
interface_name
An existing IP interface. To display interface
names, use the IP LIST INTERFACES command.
N/A

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