V. DESARROLLO CAPÍTULO 2
5.1 Comparación del sistema de alarma para diferentes vehículos
You should consider what your billing requirements are for transferred calls. Most enterprise VoIP networks have no requirement for separate billing for VoIP call legs within an internal VoIP network. Most enterprise networks are concerned only with billing for external PSTN call legs. The billing for a PSTN call leg usually goes to the party identified as the calling party on the outbound PSTN call setup. For a PSTN call using ISDN BRI/PRI, the calling party number is passed from the Cisco CME extension that originated the call. You can use the Cisco CME
dialplan-pattern or Cisco IOS voice dial peer translation rule commands to convert from
two-to-five-digit abbreviated extension numbers into a national number format acceptable to your PSTN service provider. If you are using simple analog FXO port connections to connect to the PSTN (simple subscriber line), you have no control over the billing party information, so you can probably skip the rest of this section. Calls on an analog subscriber line are simply billed to the number associated with the subscriber line by the PSTN service provider. For the ISDN PRI/BRI case, this is an area where the difference between transfer commit at connect/alerting and blind transfers may be significant. It’s also an area in which hairpin call routing may provide you with some advantages.
When an H.450.2-style transfer is committed at alerting/connected, the calling party number for the consultation call setup to the transfer-to party (from the transferor) is normally equal to the transferor’s phone number. This is usually the bill-to number that’s associated with the call. If the consultation call involves a PSTN call leg using PRI/BRI (either a direct PSTN connection on the Cisco CME router or a remote PSTN gateway call reached via an intermediate VoIP leg), it’s useful to have the initial calling party number for the outbound PRI/BRI PSTN leg equal to the transferor’s phone number. This assumes that you want any transferred call to be billed (or traceable) to the person who invokes the transfer. When the replaces operation is triggered to connect the transferee to the transfer-to party, the calling party information associated with the PSTN leg normally does not change. This means that even after the transferor has dropped out of the call, the call continues to be billed to the transferor, at least as far as the external PSTN call leg is concerned. This is true for PSTN access that’s directly on your Cisco CME router and also when the PSTN access is on a remote VoIP-PSTN gateway accessed via a VoIP link. This is because the H.450.2 call transfer replaces operation is confined to the H.323 VoIP network. The replaces operation normally cannot extend into the PSTN connection.
Transfers that use the blind mechanism work differently. In the blind transfer case, the transferor does not originate a consultation call. The initial call received by the transfer-to party in an H.450.2 transfer case by default has the transferee’s phone number as the calling party. The transferee is often a phone number belonging to some external party. You are often not permitted to bill calls to this phone number even if you want to. Your PRI/BRI PSTN connection
is very likely to reject any outbound calls that attempt to claim an external number as the calling party identifier.
You can work around this issue in a couple of different ways, depending on the reason you chose to select the blind transfer method. For example, you may be using blind transfer to avoid the H.450.2 replaces operation if it is not supported by your PSTN access voice gateway. The workaround methods include the following:
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You can place a translation rule on the dial peer associated with the outgoing PRI/BRI PSTN port that overwrites the transferee calling party number with the general public phone number for your company.•
You can elect to force hairpin VoIP routing with transfer commit-at-alerting/connect as an alternative to blind transfer such that the outgoing PSTN call carries the transferor’s phone number.To use the first alternative, you must have control of the PSTN gateway. This is true if the PSTN access is local to your Cisco CME router. This may not be true if you get remote PSTN access across a WAN connection from a VoIP telephony service provider (TSP). In this case, your VoIP TSP may share the PSTN access ports across multiple end customers.
The second hairpin case is the most robust approach, because it forces a separate call leg to be generated for the outgoing PSTN call segment.
To force hairpin VoIP call routing, you can switch on H.450.12 services on your Cisco CME router and use a separate PSTN gateway router on which H.450.12 is disabled (or not supported). Alternatively, you can explicitly turn off H.450.2 service on your Cisco CME voice dial peers that route calls to the PSTN gateway router. You do this using the no supplementary-
service h450.2 command, as shown in Example 5-38.
Because Cisco CME includes the standard Cisco IOS voice infrastructure functionality, you can also connect your Cisco CME system to a Remote Authentication Dial-In User Service (RADIUS) server to capture call records for more detailed call tracking information collected. If you don’t have a RADIUS server, you can also configure your Cisco CME system to generate SYSLOG messages that include call details. You can use a simple PC as a SYSLOG server to record the call data, using one of several freeware SYSLOG programs available on the Internet.
Example 5-38 Turning Off H.450 on a Dial Peer
router#show running-config dial-peer voice 100 voip destination-pattern 9.T session target ipv4:10.0.1.20 no supplementary-service h450.2