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This is a brief introduction on how the PSTN signalling modules are working together in Yate.

Analog cards or CAS signalling

Supports FXS and FXO interfaces, signaling is associated to each channel

  • Directly (analog card)
  • CAS signaling over TDM (channel banks)
  • MGCP gateway with line package

The analog module provides the common call control logic.

The lower level tdmcard or zapcard modules provide both voice and access to signaling data.

If MGCP is used the actual voice is transferred by the yrtpchan module.

At a minimum you have to configure the analog channel and one of tdmcard, zapcard or mgcpca modules.

The referencing is like this:

Call Controller -> Voice interfaces


Supports signaling and voice to be fully associated or not (NFAS)

  • PRI or BRI cards with ISDN D-Channel
  • D Channel can be backhauled by SIGTRAN IUA over SCTP/IP
  • ISDN voice channels on any supported technology

The lower level wpcard or zapcard modules provide both TDM voice and access to the signaling D channel.

SIGTRAN IUA can be configured in ysigchan to transport the Q.921 signaling of the D channel over SCTP/IP from a Signaling Gateway.

In this case the voice is usually transported over MGCP.

At a minimum you have to configure a Q.931 call controller in ysigchan and voice circuits in wpcard, tdmcard or mgcpca.

If SIGTRAN IUA is configured you must configure the Q.921 level separately in ysigchan and a matching transport in sigtransport else the HDLC interface must be defined in the TDM card module.

The referencing is like this:

Call Controller -> Q.921, Voice interfaces
Q.921 -> HDLC Interface (a proper default exists)
Q.921 -> IUA Client (only for IUA)
Q.921 -> Transport (only for IUA)


Supports associated or quasi-associated signaling

  • MTP2 on TDM cards
  • SIGTRAN M2UA over SCTP/IP to a Signaling Gateway
  • SIGTRAN M2PA over SCTP/IP to another SS7 node
  • Cisco SLT backhaul
  • Signaling Transfer Point functionality (optional)
  • ISUP voice channels on any supported technology like TDM and MGCP

If MTP2 is used the wpcard or zapcard modules provide access to the TDM signaling link.

SS7 Layer 2 can also be transported over SIGTRAN of over Cisco SLT.

Links are configured together in linksets described in ysigchan.

Each linkset defines part of the SS7 network topology but has absolutely nothing to do with voice.

Phone calls are signaled between SS7 points by the ISUP protocol running on top of SS7.

Sections with type=ss7-isup in the ysigchan configuration describe each instance defined by the local point code, remote point code and their circuits.

For each such ISUP instance the circuits can be provided over TDM B channels (wpcard, zapcard) or a remote Media Gateway using MGCP.

At a minimum you have to configure an ISUP call controller, a linkset (MTP3) and one or more links in ysigchan. Voice circuits are defined in wpcard, tdmcard or mgcpca.

If SIGTRAN protocols are used you must also configure matching transports in sigtransport. If Cisco SLT is used for MTP2 backhaul each link must be described in ciscosm.

The referencing is like this:

Call Controller -> Voice interfaces
Call Controller -> Point Codes (local, remote)

Linkset -> Links
Linkset -> Point Codes (local, adjacent, routes)

Link -> Interface (only if link is MTP2)
Link -> M2UA Client (only if link is M2UA)
Link -> Transport (only if link is M2PA)

M2UA Client -> Transport

Again, there is no direct association between the Call Controller and the Signaling path. The only correlation is established by the Point Codes.

If a Gateway is used it is usually a combination of Signaling Gateway (SIGTRAN, Cisco SLT) and Media Gateway (MGCP).

For both ISDN and ISUP any association between them must be established explicitly by configuration.

Of course it is possible that specific gateways are only Signaling or only Media but normally you need at least one of each.

See also

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