EndPointAbstractionInterface (SIG1)

Description

SIP abstraction of an End Point

Release notes

Show release notesHide release notes

Resource configuration interface

Resource configuration

• Use authentication: Select if you want to authenticate calls (INVITE). Register and subscribe always use authentication.
• DTMF mode: Select RFC2833 or Inband
• Password: If authentication is used indicate here the password. Limitations: when using "Subscribe to MWI", character ":" is forbidden and before version 2.6.1 of this resource, characters ":" and "/" are always forbidden.
• SIP server feature:
  • SIP reinvite: Indicate 'Default' if you want to use the parameter indicated in Communication Server, yes or no.
  • Qualify SIP devices registration: Indicate yes if you want to detect the remote peer is unavailable by sending OPTIONS messages
  • Force specific codec: Indicate yes, if you want to force codec
  • First codec: First codec advertised in SDP
  • Second codec: Second codec advertised in SDP
  • Third codec: Third codec advertised in SDP
  • Sip socket id: To use another source IP address and/or port than the default one, this field can be set to an id corresponding to the list of "IP:Port" that can optionally be defined in Communication Server Module. Leave empty if you don't need this feature. An id can also be chosen automatically using the last digits of the resource id, when using the special value automatic_min=00max=00offset=00. Offset is a positive or negative integer applied to the resource id and if the result is between min and max it is used as socket id, otherwise it won't be set. Note that this feature only works with the 10000 first SIG1 resource, i.e SIG10001 to SIG19999, not with the following SIG1A000 and higher.
  • Caller ID method (formerly Send Remote-Party-Id):
    • Use From header: This is the default header. The caller number and the caller name are sent via the the From header
    • Use Remote-Party-Id header: The caller number and the caller name are sent via the Remote-Party-Id header
    • Use P-Asserted-Id header: The caller number and the caller name are sent via the P-Asserted-Id header
  • Trust Remote-Party-Id: Indicate yes, if you expect SIP message received contains Remote-Party-Id header
  • Optional digest user name: This value can override the authentication user on the SIP server (by default the remote peer is used).
  • Send REFER on: Select on which call the REFER message should be sent in an attended call transfer scenario.
  • Early Media Suppression: Set this to yes to block all early media from being sent to the remote peer. When not set, this corresponds to the Inband Ringing setting of Communication Server. (needs CS >= 3.13)

• Abstracted Network Prefix: In order to select a specific 'ringer' service, indicate a prefix. Use the external number mapping in order to map this prefix to the ringer service. Limitations: Max 79 characters, valid directory extension.
• Abstracted Network Context: In order to select a specific 'ringer' service, indicate a context. Use the external number mapping in order to map this prefix to the ringer service. Limitations: Max 79 characters, valid restriction group.
• Remote peer: SIP endpoint identity which will be registered. Limitations: Max 79 characters (for versions < 2.0), restricted to ALPHA / DIGIT / "-" / "." / "_" (see below for more details).
• Caller ID: Caller ID of the EndPointAbstraction. Limitations: Max 79 characters, valid directory extension.
• Register:
  • Enable: Indicate if you want to register or not
  • Timeout: Set the expiration interval of the registration in seconds (the default value of Communication Server is used when the field is empty)
• SIP Server 1: IP address or hostname of the first IP/PABX.
• SIP Server 2: IP address or hostname of the second IP/PABX, used for redundancy purposes.
  • Hostname can be if the following format: SOA1XXXX.sip.dynamicpeer in which case the address and port of a SIP Trunk will be used. If the SIP Trunk address is dynamic, it will be regularly refreshed, taking into account registrations in that SIP Trunk.
• Subscribe to MWI: Indicate if you want to subscribe to the Message Waiting Indicator or not.
• Allow user control of registration flag (only in version 1.75.0, deprecated): Is the user able to change the state of the EndPointAbstraction (registered/unregistered) or not.
• Nonstandard SIP: This section exposes some settings which should not be used unless the remote end cannot be made compatible with the Standard SIP.
  • Ignore SDP version: When receiving an SDP, the session version provided in the SDP will not be taken into account and we will always completely process all the SDPs received. This is not compliant with RFC4566.
  • Symmetric Response Routing:
    • Automatic: default CS behavior, SIP compliant routing based on Via header.
    • Force rport: Behave as if the "rport" option was provided in the Via header (RFC 3581), i.e. send SIP responses to the source IP and port from which the request originated instead of the address/port listed in the top-most Via header.
  • NAPT PBX URI domain: Override the domain of the URI and To header, without impacting the real destination of SIP Requests. Useful when the IP/PABX is behind a router doing NAPT and not a real SIP proxy.

• EPA Activated: When set to no the endpoint is deactivated, as if the resource did not exist.

Specific limitations:

• All fields: Unless specified otherwise, the semicolon character ";" is not allowed.
• Remote Peer: This value will be used in SIP messages as the user part of a SIP URI. It is restricted to ALPHA / DIGIT / "-" / "." / "_" which is a known working subset of what is documented in RFC3261 (SIP), Section 19.1, which points to RFC3986 (URI), Section 3.2.1. These documents define more characters that should work, although not everything was tested. It is known that the column character ":" can not be used as-is but should work once percent-encoded, like other characters listed as "sub-delims".