Module configuration parameters
- SIP canreinvite: when SIP initiates the call, the INVITE message contains the information on where to send the media streams. Asterisk uses itself as the end-points of media streams when setting up the call. If canreinvite is set to yes, once the call has been accepted, Asterisk sends another (re)INVITE message to the clients with the information necessary to have the two clients send the media streams directly to each other
- If one of the clients is configured with canreinvite=NO, Asterisk will not issue a re-invite at all.
- If the clients use different codecs, Asterisk will not issue a re-invite.
- DNS Lookup: Allow asterisk DNS entries manager
- SIP videosupport: Configure asterisk to support video
- Log Level:
- Indications: tone localization.
- Default Context: default context for SIP devices (blank=default allow unauthenticated SIP devices to call internally only)
- IAX Jitter Buffer: enable or disable IAX jitter buffer. Enabling jitter buffer is useful when you have quality issues with IAX trunks over internet
- Qualify SIP devices registration: Default qualify configuration for SIP devices. If you turn on this option, the process will send SIP OPTIONS messages every 60s. If the device does not respond within the defined delay, it will be considered as off-line. Selecting 'no' will disable this functionality. Selecting 'yes' will use a good default timeout value (2000 ms). A custom timeout delay (expressed in milliseconds) can also be defined.
- Voicemail access Language and Type: The default voicemail language and possibilities.
- English-Default : An advanced voicemail system with English audio messages.
- French-Default : An advanced voicemail system with French audio messages.
- French-Lite : A simple voicemail system with French audio messages.
- EscauxTsp secret: Secret needed by the Microsoft Telephony Service Provider plugin installation
- Voicemail email sender: Sender address of voicemail emails. If no domain specified, will be suffixed by the Domain name defined in Mail Server module. Default: noreply.
- WAV file voicemail email compression:
- Yes: attached wav file will use compression (NB: Some Windows Media Player versions do not support wav compression correctly).
- No: attached wav file won't use compression.
- Disabled: no audio file will be attached to email.
- Server side DSCP marking: enter a comma separated list of 4 DSCP values for SIP, RTP audio, RTP video, IAX. Possible values are listed in the Annex.
- mISDN support: default value is 'no'. Select 'yes' if you wish additional support for mISDN type of ISDN cards.
- Auto-cleaning of unreachable SIP devices: When enabled, force cleaning of hanging channel if SIP peer is unreachable. Use this option if you detect hanging channels with phone that are not reachable any more, typically the softphone. The option requires the feature ' Qualify SIP devices registration ' to be enabled in order to detect unreachable SIP phones.
- Number of park slots: Specify the number of park slots available. Typically this is set to 5 times the number of netConsole.
- Parktime: Specifies the maximum park time in seconds. If a net.Console is used, it should be set to 3600.
- In-band ringing: When set to 'Never', never generate an In-band ring tone, when set to 'yes', always generate in-band ring tone, when set to 'no' activate old behavior with out-of-band ringing followed by in-band ringing. The recommended value is 'never'. If in some case no ringing tone is heard, it can be set to 'yes'.
- Group-pickup extension: set the extension to be used to pick up a ringing phone of the same pickup-group
- Maximum call duration: set a global maximum call duration for any call. The time is to be set in seconds and its default value is one day (86400 s).
- Asterisk console timeout: set a maximum time an Asterisk console can be open. The time is to be set in seconds and its default value is one day (86400 s).
- Blind transfer key: allows caller or called device to transfer ongoing call to another party (internal extension or external number). Please consider this function as very unsecure and use it at your own risks! Use option '#' for normal operation, 'custom' is special for Escaux FMU. Linked global variable: DialAsteriskOption.
- The following options are available with SMP >= 4.7.0
- Voicemail to email click-to-dial: set to enable if you want to add a link in the emails that you receive when a message is left in your voicemail. This link will transfer you to a service to hear your message.
- Voicemail to email click-to-dial service (default: 8500): set the number to be called to perform the service mentioned above. Parameters available for this service are : msgcid (the caller), msgcalledid (the user called), msgid (message id in the mailbox), msgdur (message duration). The number entered will be written in the mail sent to the user.
- Voicemail to email click-to-dial call a phone service (default: user's extension) : indicate here the number to use to reach the voicemail owner. By default, it will call its extension. The parameters available to create the service are the same as for the previous option.
- Sync Extension Policy (default: Minimum): Specify the action allowed in the context of an apply-extension-change. Set to 'Minimum' to only allow SIP reload if a phone has been added. Set to 'AllowVoicemailReload', to allow voicemail reload if an extension has been added or modified. The settings does not concern an apply-sop-change or an apply-cluster-change.
- Disable IAX on external interfaces (default: no): Allows you to select if you want to bind to the IAX service to external interface. 'no' if you want to bind to localhost, 'yes' if you want to bind it to the external interface. If you do not use IAX, then you can set this option to 'yes'.
- Disable AMI on external interfaces (default: no): Allows you to select if you want to bind to the Asterisk Management Interface to external interface. 'no' if you want to bind to localhost, 'yes' if you want to bind it to the external interface. Since SOP API 4.0.0+ you can use 'yes' otherwise you must use 'no'.
- Codec negotiation (default: independent): Define how the codec negotiation will occur. If "independent" is chosen, the codecs will be negociated independently for each channel of the communication. As a consequence, transcoding may often happen while there is a common codec between the peers. If "followpeer" is chosen, Asterisk will try to match the codecs of the peer initiating the call with the codecs of the peer receiving the call. This does not ensure that a common codec is selected in case of transfers or call pickups, but it should decrease the frequency of the transcoding when matching codecs are present. Limitations apply.
- Codec to use: It will create a codecs list with priority to use for the whole system. Limitations apply.
- Voicemail maximum length message (in minute): This will set the maximum length in minute of an incoming message. By default, it is set to 30 minutes.
- Voicemail notification (by email) language (default : english): Allow you to configure notification language. It can be configured to send to e-mail in several languages.
- Activate core dump: This option indicates of core dump should be created by the system in case of process abort. Default is 'yes'.
- Always fake user rejection: When an incoming INVITE or REGISTER is to be rejected, for any reason, always reject with an identical response equivalent to valid username and invalid password. This reduces the ability of an attacker to scan for valid SIP usernames. Default is 'yes'.
- Allow guest calls: Allow calls to the Default Context without any authentication. Important when listening on a public IP. Default is 'yes'.
- Email notification for new voicemail (default: yes): This option enable the notification sent by email when a voicemail is received. Note that a restart of Asterisk is needed to take the new value into account.
- Enable enriched Remote Party Presence Information: This option enables the Remote Party Presence feature. Default is 'no'. See Remote Party Presence for more information.
- Email Return Path: Set the return-path header to emails sent by asterisk
Asterisk has to be restarted after the module installation. This can be done through the SOPShell
Navigate to: Diagnostics > Telephony > asterisk console
Type in the following command:
00000XXX*CLI> restart now
- Check dependencies
In case this module is installed on a Baseline 1 High Availability SOP that is currently in standby mode, you need to use the shell plugin to deactivate the processes. This is available in the High Availability module version > 2.6.0.
Multi-codec VoIP environments can become very complex, with each endpoint having a specific implementation of the codec negotiation algorithm. While basic multi-codec environments are supported with this module, some use cases might lead to unexpected results. We therefore recommend using the successor of this module: Communication Server
Specifically, in an environment that mixes phones with several codecs and phones that support only one codec, the sound may be deformed on one side after some transfer scenarios. If this problem arise, change the "codec negotiation" option from "followpeer" to "independent".
Annex: Diffserv/DSCP values
| Diffserv traffic class
|| Hex value
|| Dec value
For more information see:
Note : This module is using a modified version of Asterisk project. Please contact D4SP Support (firstname.lastname@example.org
) to request the source code