Channels
From Taridium
Channels specify how an end-point communicates with ipbx.
SIP Channels
The SIP Channel Module enables ipbx to communicate via VOIP with SIP telephones and exchanges. These are usually SIP clients. Trunks and other static end-points are usually configured separately.
The SIP channels interface lists all SIP channels that are configured on the system. The system will automatically update the IP address and the expiry time on dynamically registered channels. Move your mouse pointer over the expiry time to see more details about the registration.
Adding a SIP Channel
In order to add a SIP channel click on the add channel button at the bottom of the page. Taridium's ipbx supports a powerful templating engine that comes pre-loaded with common SIP phones.
Click on load template to load the template settings and change the other settings accordingly.
In order to create the appropriate text configuration file four auto-provisioning you will need to either select the MAC address of the phone or select it from your DHCP server. Unused MAC addresses will show up in 'green'. See also System → Devices and Channels → SIP Channels → Templates.
Note that the MAC address needs to be entered when the channel is created and cannot be applied retroactively!
Accountcode
Users may be associated with an accountcode. Keep ‘default’ selected if you are not sure. This must be of the format [mailbox number]@[mailbox context] for complete end-user panel functionality.
Allow Codec
Allow codecs in order of preference (Use Disallow = all first, before allowing other codecs) disallow Disallow all codecs for this peer or user definition.
Amaflags
Categorization for CDR records. Choices are default, omit, billing, documentation.
Caller ID Name & Caller ID Number
Caller ID information used when nothing else is available.
Callgroup
num1,num2-num3 : Defines call groups for calls to this device.
Can Re-invite
If the client is able to support SIP re-invites. Default yes.
Context
If type=user, the Context for the inbound call from this SIP user definition. If type=peer, the Context in the dialplan for outbound calls from this SIP peer definition. If type=friend the context used for both inbound and outbound calls through the SIP entities definition. If no type=user entry matches an inbound call, then a type=peer or type=friend will match if the hostname or IP address defined in host= matches. This is usually set to inside.
Default IP
Dotted.Quad.IP.Addr : Default IP address of client host= is specified as DYNAMIC. Used if client has not been registered at any other IP address. Valid only for type=peer.
DTMF Mode
inband|info|rfc2833 : How the client handles DTMF signalling. Default rfc2833.
From User
Specify user to put in "from" instead of callerid (overrides the callerid) when placing calls _to_ peer (another SIP proxy). Valid only for type=peer.
From Domain
Sets default From: domain in SIP messages when placing calls _to_ peer. Valid only when in [general] section or type=peer.
Host
dynamic|hostname|IPAddr : How to find the client - IP # or host name. If you want the phone to register itself, use the keyword dynamic instead of Host IP.
Incoming Limit and Outgoing Limit
Number : Limits for number of simultaneous active calls for a SIP client. Valid only for type=peer.
Insecure
very|yes|no|invite|port : Specifies how to handle connections with peers. Default no (authenticate all connections).
Language
A language code defined in Indications - defines language for prompts
Mailbox
Voicemail extension (for message waiting indications). Valid only for type=peer.
MD5 secret
MD5-Hash of "<user>:ipbx:<secret>" (can be used instead of secret)
Nat
yes|no : This variable changes the behaviour of ipbx for clients behind a firewall. This does not solve the problem if ipbx is behind the firewall and the client on the outside.
Permit and Deny
IP address and network restriction
Pickup Group
Group that can pickup fellow workers' calls using *8 and the Pickup() application on the *8 extension
Port
SIP port of the client
Qualify
yes|no|milliseconds : Check if client is reachable. If yes, the checks occur every 2000 milliseconds (2 seconds). Valid only in [general] section and type=peer.
Restrict CID
(yes/no) To have the callerid restricted -> sent as ANI; use this to hide the caller ID.
RTP Timeout
(seconds) Terminate call if x seconds of no RTP activity when we're not on hold. Valid only in [general] section and type=peer.
RTP holdtimeout
(seconds) Terminate call if x seconds of no RTP activity when we're on hold (must be larger than rtptimeout). Valid only in [general] section and type=peer.
Secret
If ipbx is acting as a SIP Server, then this SIP client must login with this Password (A shared secret). If ipbx is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, then this field is used to authenticate SIP INVITEs that ipbx sends to the remote SIP server.
Type
user|peer|friend: Relationship to client - outbound provider or full client?
Username
<username[@realm]> : If ipbx is acting as a SIP client to a remote SIP server that requires SIP INVITE authentication, then this field is used to authenticate SIP INVITEs that ipbx sends to the remote SIP server. Also, for peers that register with ipbx, this username is used in INVITEs until we have a registration.
IAX Channels
The IAX Channel Module enables ipbx to communicate via VOIP with IAX telephones and exchanges. Only IAX2 communication is supported at this point. Trunks and other static end-points are usually configured separately.
