Device settings

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Screenshot

Device settings sip.png

Each device can have many settings. Screenshot shows settings for SIP device. These settings are similar to other device types also.

There are several groups of settings: General, Authentication, CallerID, Network Related, Groups, Voicemail, Codecs, Allowed Addresses, Advanced, Tell Options, Debug and Recordings.

General

  • Accountcode – unique id of the device in the system
  • Description – for informational purposes
  • PIN – device PIN for authentication/authorization
  • Device group - to which group device belongs (not used most of the time)
  • Type - what type is this device
  • Extension – short number by which this device can be reached - must be unique in the system
  • DTMF Mode – available options: inband, info, RFC2833, auto – choose which one your provider uses (RFC2833 most used)
  • Works not logged – does user need to log to MOR GUI to be able to dial out? (mainly used for Call Center environments, leave it default in most cases)
  • Location – default Global, choose appropriate one based on Localization needs
  • Timeout - how long to try to reach this device and when hangup if device does not answers
  • Trunk – No/Yes/Yes with ANI – Is this device Trunk with/without ANI?
  • Call Limit - how many simultaneous device user can make?

Authentication

For ZAP devices:

  • Channel – which channel (or channel group) to use on PRI/BRI/PSTN card. Channels and groups should be configured in zapata.conf.

For all other device types:

  • Username – username you enter in your device
  • Password – password you enter in your device
  • Authenticate by IP – should it be authenticate by IP (not by username/password)?


CallerID

CallerID – consists of two parts: name and number. The "Number" part gets transferred by default by all technologies: GSM, PSTN, SIP etc, the "Name" part only on some. The number you see on your mobile phone when somebody is calling you is "Number" part.

  • Name – the "Name" part of CallerID
  • Number – the "Number" part of CallerID

NOTE: if you leave these fields empty – user can set CallerID by himself. Usually is not a good way to let it do it for user.

  • Number from DID - put DID as CallerID (only availale when device has DIDs)
  • CID control by DIDs – only available when this device has DID(s)

This setting let's you control what CID user can enter in his devices. These CID numbers should be from the set of device's DIDs. It is mainly used when user's PBX is connected over trunk and many DIDs are routed to this Trunk. Calls coming out from this PBX must have CallerID Number from the set of DIDs assigned to this Trunk (PBX). If CID from DIDs is checked - system checks if device's cid number is from dids assigned to this device. If no dids are assigned - this value is ignored. In order for this setting to be active - CID Name/Number fields must be empty to allow user enter any CID number he wants. If CID from DIDs is checked and user enters any CID (not from his DIDs) - system will change his CID by his first available DID (selected from database by lowest id).


Network Related

  • Host – device IP or hostname, the value can be "dynamic", which means that device should register (it should be like this most of the time).
  • IP Address - should be devices IP or disabled if device is "dynamic"
  • Port – port used to connect to device (default: 5060 for SIP, 4569 for IAX2). If host = dynamic this field is updated by Asterisk when device registers. It is port from where call is coming (from device).
  • Last time registered - when device was last time registered to the Asterisk server
  • Media control – canreinvite/transfer – do you want your server to stay in media path between your clients and your provider
  • NAT – available options: yes, no, never, route. For detailed explanation about these settings please refer here.
  • Qualify – how often "ask" device about availability. More details here for SIP devices and here for IAX2 devices.


Groups


Voicemail

  • Email – to which send received Voicemail
  • Password – digit password user should enter when he calls Voicemail number to listen for messages


Codecs

Choose the codecs your provider uses.

NOTES:

  • When no fields are checked – that means that all codecs are available. E.g. settings in sip.conf or iax.conf are effective.
  • If Provider and Device does not have similar codecs - call can not be established.


Allowed Addresses

IP, MASK – permit traffic from these IP only. You can find detailed explanation here.

If you do not clearly understand why these settings are used for - leave them with default values.


Advanced

  • Fromuser/Fromdomain – This is used when calling TO this peer FROM asterisk. If you're using _register=>_ with another SIP proxy, this setting can come in handy since some SIP networks only allow users in the right domain with the correct user name.
  • Trustrpid – This defines whether or not Remote-Party-ID is trusted. It's defined in http://tools.ietf.org/id/draft-ietf-sip-privacy-04.txt
  • Sendrpid – Defines whether a Remote-Party-ID SIP header should be sent. Defaults to no. This field is often used by wholesale VoIP providers to provide calling party identity regardless of the privacy settings (the From SIP header).
  • Insecure
    • port: ignore the port number where request came from
    • invite: don't require authentication of incoming INVITEs
    • port,invite: don't require initial INVITE to authenticate and ignore the port where the request came from
  • T.38 support - Asterisk does not have good T.38 support so use this option just for testing
  • SIP 302 Reinvite support - SIP Reinvite support for device, disabled by default
  • Progressinband:
    • yes - When "RING" event is requested, always send 180 Ringing (if it hasn't been sent yet) followed by 183 Session Progress and in-band audio
    • no - Send 180 Ringing if 183 has not yet been sent establishing audio path. If audio path is established already (with 183) then send in-band ringing (this is the way asterisk historically behaved because of buggy phones like polycom)
    • never - Whenever ringing occurs, send "180 ringing" as long as "200 OK" has not yet been sent. This is the default behaviour of Asterisk.
  • Video support – does your provider support Video over IP? More info here.
  • Allow duplicate calls - default no


Tell Options

  • Tell balance – should MOR tell balance to user when he is trying to dial (every time) – default no
  • Tell time - should MOR tell remaining time to user when he is trying to dial (every time) – default no
  • Tell remaining time when left – when some time is left MOR will tell remaining time to talk (in seconds)
  • Repeat remaining time when left – repeats remaining time when left some time (in seconds)


Debug

SIP debug info - Can be enabled per SIP device basis. It is useful in debugging. And it is saved to calls table when it is enabled.


Recordings

This section is available when Recordings Addon is installed in the system.

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