SIP trunk configuration is divided into basic, advanced and codec configuration, accessible from their accompanying tabs on the SIP Provider form.
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Field |
Content |
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Outbound Caller ID |
Optional caller ID number to present to the remote party when placing outbound calls over the trunk |
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Outbound Dial Prefix |
Optional prefix to automatically prepend to dialed numbers for outbound calls. |
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Name |
The unique name of this SIP trunk. If this trunk is a SIP provider, the name of the trunk must match the username. |
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Description |
A custom description of this SIP trunk |
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SIP Server |
The address of the SIP provider peer |
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Port |
The port on which the SIP service resides (defaults to 5060) |
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Username |
The registration user name provided in the registration string to the SIP provider when registering. When registering, the username must be the same as the username configured in the SIP provider. When not registering, if “Identify by username” is enabled, this username is used to match incoming SIP calls. I.e. the from-user field for the remote peer must be set to this value. |
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Password |
The registration password required by the SIP provider |
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Act as Provider |
When enabled, this trunk allows other SIP users to register with it. An UNKNOWN peer will be created and the IP address of the peer will be associated with this trunk on registration.
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Register with Provider |
When enabled, the SIP trunk will attempt to register its IP address with the SIP host at the provider address configured. The registration string will use the configured username. |
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Identify by username |
When enabled, will use the username field to identify a SIP client. This is necessary when multiple SIP trunks are registered with the same SIP provider from the same IP address and allows the provider to determine the incoming trunk from the SIP FROM field provided. The remote SIP peer’s from-user field needs to match this trunk’s name and username field if enabled. |
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Field |
Content |
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Proxy |
The SIP proxy to use when establishing outgoing connections in peer to peer setups. This will in most cases be left blank when registering with a service provider, although some service providers may require registration via a SIP proxy. |
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Proxy port |
The port on which the SIP proxy service resides (defaults to 5060) |
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From user |
Defaults to the extension number of the initiator of the SIP call. If the provider rejects the registration, the from user might need to be set to the provider’s registration / account / username. The from-user field only affects SIP calls, and not SIP trunk registration. This fields overrides the caller ID number sent in the SIP call setup headers. |
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From domain |
A domain to append to the from-user during call setup. I.e. from-user@from-domain. |
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Auth user |
An authentication username to use during SIP registration and call setup. |
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NAT |
When enabled, the system ignores the SIP and SDP headers’ address and port and replies to the sender’s address and port. This should be enabled when the Com.X is located behind a NAT router. |
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Secure |
If this is setting is enabled, after registration with the SIP provider, all invites will require successful authentication as well as port number matching. Disabling this settings allows invites to proceed without further security requirements after successful registration. |
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Send Remote ID |
Defines whether a remote party ID SIP header should be sent to the peer. This SIP header is often sent by SIP providers to provide calling party identity regardless of the privacy settings on the trunk. |
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Trust Remote ID |
If enabled, then the Remote Party ID SIP header will be used to determine the calling number for an inbound call rather than the From header. |
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Retry registration |
If network problems result in trunk intermittently becoming unavailable, enable this option in order to keep retrying registration every 30 seconds, indefinitely. If this option is disabled, registration attempts will stop after 10 failed attempts. |
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Qualify connection |
When enabled, checks the reachability of the peer every 60 seconds.
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Contact # for incoming |
A DID number to assign to incoming calls on this SIP trunk. This facilitates inbound routing where the DID is not provided by the SIP provider. |
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Maximum # channels |
The maximum number of simultaneous SIP channels supported on this trunk. Leave blank for no limit. |
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T.38 fax mode |
Configures the trunk as a T.38 end-point, allowing faxes to be sent and received using UDP over IP using the T.38 fax protocol. Also configures UDPTL and datagram settings accordingly. |
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Max datagram size |
Fine-tuning for T.38 |
Codecs supported by a SIP trunk can be selected by using the Codecs button. Use Ctrl-Left-Click (holding Ctrl down) to select multiple codecs. Individual codecs can be moved up and down the codec priority list by selecting the codec entry, and selecting the Up or Down buttons.
When negotiating a SIP call, the codec with the highest priority (at the top of the list) will be attempted first, then the second and so on. Codecs may specify the integration time in milliseconds after the codec name in the form codec:integration time, e.g. ‘g729:40’
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Note: Com.X systems support dynamic RTP payloads in the range 96-127. These are set up per-call by the peer using the SDP a:rtpmap parameter. |