Username

User that this SIP user is associated with.

SIP user

Username used to login in a Softphone or any SIP device.

SIP password

Password to login in a Softphone or any SIP device.

CallerID

The Caller ID number that will be shown in their destination. Your trunk needs to accept CLI.

Alias

Alias to dial between sip accounts from the same AccountCode (company).

Disallow

Disallow all codecs and then select the codecs available below to enable them to the user.

Codec

Select the codecs that the trunk will accept.

Host

Dynamic is an option that allows the user to register their account under any IP. If you want to authenticate the user via IP, put the client IP here, let the password field blank and set it to “insecure” to por/invite in the Aditional Informations tab.

Group

When sending an call from DID, or campaign to a group, will be called all SIP users that are in the group. You can create the groups with any name.

Is used as well to capture calls with *8, need to configurate the option “pickupexten = *8” in the file “feature.comf”.

Videosupport

Activate video calls.

Block call regex

Block calls using REGEX. To block calls from phones, just put it ^55\d\d9. You can see more details at the link https://regex101.com..

Record call

Record calls of this SIP user.

Tech prefix

Useful option for when it’s necessary to authenticate more than one client via IP that uses the same IP. Common in BBX multi tenant.

CNL zone

CNL zone used for Brazilian numbering and routing rules for this SIP user.

Description

Optional description to identify this SIP user in reports and administration screens.

NAT

Nat. You can see more details at the link https://www.voip-info.org/asterisk-sip-nat/.

Directmedia

If enabled, Asterisk tries to redirect the RTP media stream to go directly from the caller to the callee.

Qualify

Sends SIP OPTIONS packets to verify whether the user is online.
Syntax:

qualify = xxx | no | yes

XXX is the number of milliseconds used as the timeout. If the value is “yes”, Asterisk uses the time configured in sip.conf. The common default is 2 seconds.

When qualify is enabled, Asterisk sends OPTIONS packets regularly to verify whether the device is still online.
If the device does not answer within the configured time, Asterisk considers the device offline for future calls.

This status can be verified with the “sip show peer XXXX” command. Asterisk only shows qualify status when the peer has qualify enabled.

Trunk groups

:::::::WARNING::::::. By selecting a trunk group here, the trunk group will be ignored from tariffs and this trunk group will always be used. Only select a trunk group here if you really want all calls from this SIP user to be sent to this trunk group

Context

This is the context that the call will be processed, “billing” is the standard option. Only change configuration if you have knowledge of Asterisk.

Dtmfmode

DTMF mode used by this SIP user. You can see more details at the link https://www.voip-info.org/asterisk-sip-dtmfmode/..

Insecure

This option must be “NO” when the host is dynamic. For IP authentication, change it to port,invite.

Deny

You can limit SIP traffic of a determined IP or network.

Permit

You can allow SIP traffic of a determined IP or network.

Type

Default type is “friend”, which allows the SIP user to make and receive calls. You can see more details at the link https://www.voip-info.org/asterisk-sip-type/..

Allowtransfer

Allows this VoIP account to transfer calls. The transfer code is *2 plus the extension. Asterisk must have atxfer => *2 configured in features.conf.

Fake Ring

Activate false ring. Add rR of the “Dial” command.

Call limit

Maximum simultaneous calls allowed for this SIP user.

MOH

Waiting music for this SIP user.

URL events notify

.

Addparameter

The parameters set in here will replace the system default parameters, as well of the trunks, if there’s any.

AMD

.

Forward type

Resend destination type. This resend will not work in queues.

IVR

IVR that will receive the call if this SIP user does not answer.

Queue

Queue that will receive the call if this SIP user does not answer.

Sip user

SIP user that will receive the call if this SIP user does not answer.

Destination

Click for more details
We have 3 options, conform the selected type, group, number or custom.

* Group, the group name set here, needs to be exactly the same group of SIP users that wants to receive the calls, is going to call all SIP users in the group.
* Custom, it’s possible to execute any valid option of the DIAL command of Asterisk, example: SIP/contaSIP,45,tTr
* Number, can be a landline number or mobile number, needs to be in the 55 DDD format

Dial timeout

Timeout in seconds to wait for the call to be picked-up. After the timeout will be execute the channeling if it’s configured.

Enable voicemail

Activate voicemail. It’s necessary the configuration of SMTP in Linux to receive the email with the message. You can see more details at the link https://www.magnusbilling.org/br/blog-br/9-novidades/25-configurar-ssmtp-para-enviar-voicemail-no-asterisk.html..

Email

Email that will be send the email with the voicemail.

Password

Voicemail password. It’s possible to enter in the Voicemail typing *111

Parameters

Additional SIP parameters written for this account. Use only valid Asterisk SIP options.

Peer

sip show peer

Forward type

Type of forwarding applied when this SIP user does not answer or is unavailable.