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C H A P T E R6Connecting Multiple Cisco Unified CallManagerExpress Systems with VoIPThis chapter describes the ways in which you can use Cisco Unified CallManager Express(Cisco Unified CME) as a component of a larger network using the two major Voice over IP (VoIP)protocols—H.323 and SIP—to link multiple Cisco Unified CME systems. It examines some of theconsiderations that apply within a networked environment that do not arise in simpler standaloneconfigurations. This chapter focuses on the call handling implications of using Cisco Unified CME in anetwork.The following sections address specific multiple Cisco Unified CME deployment issues:Note Considerations When Integrating Cisco Unified CME in H.323 and SIP VoIP Networks, page 6-1 Integrating Cisco Unified CME in an H.323 Network, page 6-4 DTMF Relay for H.323, page 6-17 Call Transfer and Call Forwarding in an H.323 Network Using H.450 Services, page 6-20 Integrating Cisco Unified CME in a SIP Network, page 6-30For additional information, see the “Related Documents and References” section on page xii.Considerations When Integrating Cisco Unified CME in H.323and SIP VoIP NetworksH.323 is the dominant protocol deployed for VoIP networks from an installed-base perspective. BecauseH.323 is more mature than SIP, you can expect to see increased real-world interoperability betweendifferent vendors’ H.323 products, particularly with basic call handling. However, many of thehigh-level VoIP networking considerations that apply to H.323 apply equally in the SIP context. Sometechnical and protocol-specific differences exist between H.323 and SIP VoIP networking, but for themost part, you’ll find more commonality than difference, at least at the level of technical detail that thischapter addresses.The shared aspects of the two protocols means that the overall high-level architecture and distributionof hardware and primary component roles within your VoIP network don’t significantly depend on whichprotocol you choose to use for intersite VoIP. For networks built on either H.323 or SIP, you are dealingCisco Unified CallManager Express Solution Reference Network Design GuideOL-10621-016-1
Chapter 6 Connecting Multiple Cisco Unified CallManager Express Systems with VoIPConsiderations When Integrating Cisco Unified CME in H.323 and SIP VoIP Networkswith peer-to-peer communication between sites. Therefore, you also need some kind of telephonenumber directory system to be able to resolve the IP address of the appropriate destination VoIP peerdevice for intersite calls.In contrast, this similarity between H.323 and SIP does not extend to Media Gateway Control Protocol(MGCP) (and also Skinny Client Control Protocol [SCCP]), which takes a significantly differentapproach to telephony. Of course, it is still possible to connect Cisco Unified CME to MGCP networks,primarily using either H.323 or SIP. Many MGCP Call Agent implementations (using MGCP internallyfor phone control) use H.323 or SIP to connect separate Call Agents (as intersystem peer-to-peer).Cisco Unified CME itself does not support control of MGCP endpoints. Cisco Unified CME uses SCCPfor phone control, and SCCP shares many common traits with MGCP.The term VoIP here specifically describes “long-distance” VoIP telephone calls that traverse a WAN.This interpretation excludes SCCP used to control local IP phones. Although SCCP technically does useVoIP technology, it is primarily used in the context of operating voice calls within the confines of a LANwith more or less unlimited bandwidth and many fewer concerns about security.You can view the H.323/SIP versus SCCP contrast as the difference between interbranch office voicetraffic and intrabranch office voice traffic, or alternatively as long distance (WAN) versus local VoIP(LAN). This division is useful in many ways, because it inherently supports the often-necessarydifference in treatment of calls between internal and external phone users.In some cases, you might want to treat H.323 calls as internal calls and not want a high degree ofdifferentiation in the treatment of LAN versus WAN calls, such as calls between separate systems on twofloors of the same building. Cisco Unified CME has features that address this, although currently youcannot treat a network of many Cisco Unified CME systems as if they are a single logical entity with fullintersite feature transparency. Both H.323 and SIP still have obstacles to overcome before this is reallypossible. Not least of these are issues surrounding meaningful interoperability with another company’sdevices for services beyond basic calls.When you extend VoIP calling into the WAN space, you might also have to consider the differencebetween VoIP calls that come from other Cisco Unified CME nodes within your WAN network versusVoIP calls that are from VoIP Public Switched Telephone Network (PSTN) gateways or even from otherindependent external/wholesale VoIP carrier networks. You can link independent VoIP networkstogether and into your corporate VoIP network using IP-to-IP gateways. This arrangement may bedesirable if you want to obtain international and long-distance phone service directly from a carrier-classVoIP service provider and have this linked at the VoIP level to your private enterprise VoIP network.NoteFor more information about Cisco IP-to-IP Gateway functionality, see the document ps5640/products qanda item09186a00801da69b.shtmlSIP potentially has some advantages over H.323 in terms of separating intersite VoIP calls from trueexternal VoIP calls, because SIP uses the Internet concept of domains. It is a fair assumption that all ofthe intersite calls will use the same root domain name and that this fact can be used to make the requireddistinction. However, from a purely practical security point of view, you will probably want any trulyexternal VoIP traffic entering your corporate VoIP network to pass through an IP-to-IP gateway and alsoa firewall, regardless of whether you choose to use SIP or H.323. This means that you should have theopportunity to appropriately classify and mark the external calls at the point of entry in either type ofnetwork.Alternatively, you can keep your VoIP network entirely separate at the IP level and simply connect intoVoIP service provider carrier networks through time-division multiplexing (TDM)-based PSTN-likegateways (at some cost in terms of increased end-to-end voice path delay). For the sake of simplicity andclarity, the rest of this chapter ignores the IP-to-IP possibility and includes only the PSTN gatewayCisco Unified CallManager Express Solution Reference Network Design Guide6-2OL-10621-01
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPConsiderations When Integrating Cisco Unified CME in H.323 and SIP VoIP Networksscenario. For many reasons, what is on the far side of the gateway—whether PSTN or IP-to-IP—is nothugely significant. It’s the gateway’s job to take care of whatever adaptation is needed to provide theinterconnection path.Cisco Unified CallManager Express Solution Reference Network Design GuideOL-10621-016-3
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 NetworkNoteWhen you configure SIP or H.323 on a router, the ports on all its interfaces are open by default. Thismakes the router vulnerable to malicious attackers who can execute toll fraud across the gateway if therouter has a public IP address and a public switched telephone network (PSTN) connection. To eliminatethe threat, you should bind an interface to private IP address that is not accessible by untrusted hosts. Inaddition, you should protect any public or untrusted interface by configuring a firewall or an accesscontrol list (ACL) to prevent unwanted traffic from traversing the router.Cisco Unified CME uses the standard ports summarized in Table 6-1 for call signaling, and mediatransport. The same ports are used by Cisco Unified CallManager and Cisco IOS voice gateway products.Table 6-1Cisco Unified CME VoIP Port NumberingProtocolPort NumbersPort TypeH.225 (call signaling)1720TCPSIP5060UDP/TCPRTP16384 to 32768UDP (dynamic)RTP (LAN)2000UDPSCCP2000TCPH.24511000 to 11999TCP (dynamic)H.225 RAS1719UDPUnicast GK Discovery1718UDPMulticast GK Discovery223.0.1.4UDPIntegrating Cisco Unified CME in an H.323 NetworkThere are two basic approaches to connecting a Cisco Unified CME system to an H.323 network: thefirst uses no gatekeeper (GK), and the second does. A direct interconnection of sites with H.323 impliesthat each site must be knowledgeable about how to reach every other site. This works well in smallnetworks of only a handful of nodes, but as the network grows larger, the configuration becomesincreasingly cumbersome to maintain. In its simplest form, a gatekeeper is a device that provides adirectory service that translates a telephone number into an IP address. Using a gatekeeper providessignificant scalability by centralizing the interconnection of the individual sites so that each site needsto be aware of only the gatekeeper and not every other site in the network.The following sections discuss different approaches to building H.323-based Cisco Unified CMEnetworks: A Simple Two-Node Topology with H.323, page 6-5 A Large Multinode Topology with H.323, page 6-7 The Role of an H.323 Gatekeeper, page 6-9 Public and Internal Phone Numbers in an H.323 Network, page 6-13 Registering Individual Telephone Numbers with a Gatekeeper, page 6-15 Internal and External Callers for VoIP, page 6-16Cisco Unified CallManager Express Solution Reference Network Design Guide6-4OL-10621-01
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 NetworkRather than being alternative approaches, they represent a simpler approach for smaller networks withonly a few nodes and a more scalable approach for larger multinode networks.A Simple Two-Node Topology with H.323In the simplest case, you can just connect two Cisco Unified CME systems via an IP-enabled serial datalink (or Ethernet), and configure VoIP dial peers on each system to symmetrically direct calls that aredestined for nonlocal extension numbers to the other Cisco Unified CME system. In other words, if theCisco Unified CME recognizes that the extension number being dialed is not present in its internal listof phone numbers, it can assume that it should send the call to the other Cisco Unified CME, as shownin Figure 6-1.Simple Two-Node Cisco Unified CME H.323 NetworkCisco Unified CME 1Cisco Unified CME igure 6-1Figure 6-2 and Figure 6-3 present flow diagrams illustrating proxy behavior betweenCisco Unified CME nodes in the two-node H.323 network illustrated in Figure 6-1.For VoIP across the WAN, all skinny and H.323 call control packets are proxied by the IP source addressof the local Unified CME router. See Figure 6-2.Cisco Unified CME VoIP Call Flow—Call Control Packet Proxy BehaviorSCCP packets from x.10.1.10to 10.1.1.1 TCP port 2000H323 or SIP control packetsfrom 10.1.1.1 to 10.1.2.1SCCP packets from 10.1.2.1TCP port 2000 to x.10.2.10IPIP phone Ax.10.1.10149944Figure 6-2IPCisco CME 1IP source address:10.1.1.1Cisco CME 2IP source address:10.1.2.1IP phone Bx.10.2.10After call signaling is established, RTP/UDP media traffic will be proxied by IP source addressUDP/RTP port 2000 of the local Unified CME router. See Figure 6-3.Cisco Unified CallManager Express Solution Reference Network Design GuideOL-10621-016-5
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 NetworkCisco Unified CME VoIP Call Flow—RTP/UDP Traffic Proxy BehaviorVoice packets fromx.10.1.10 UDP/RTPport 16384 -49152 to10.1.1.1 UDP/RTP port 2000Voice packets repacketizedand forwarded from 10.1.1.1UDP/ RTP port 16384 -49152to 10.1.2.1 UDP/RTPport 16384-491522Voice packets from10.1.2.1 UDP/RTPport to x.10.2.10 UDP/RTPport 16384 -49152IP149945Figure 6-3IPIP phone Ax.10.1.10Cisco CME 1IP source address:10.1.1.1Cisco CME 2IP source address:10.1.2.1IP phone Bx.10.2.10The following examples show the relevant configuration extracts of the two systems. It shows a pair ofCisco Unified CME systems that have extensions 1000 to1099 on Cisco Unified CME 1 (IP address10.1.1.1) and 2000 to 2099 on Cisco Unified CME 2 (IP address 10.1.2.1). Cisco Unified CME 1dial-peer voice 2000 voipdestination-pattern 20.session target 10.1.2.1dtmf-relay h245-alphanumericcodec g729r8no vadtelephony-serviceip source-address 10.1.1.1 port 2000 Cisco Unified CME 2dial-peer voice 1000 voipdestination-pattern 10.session target 10.1.1.1dtmf-relay h245-alphanumericcodec g729r8no vadtelephony-serviceip source-address 10.1.2.1 port 2000NoteThe dtmf-relay configuration portion of the output is explained in the “DTMF Relay for H.323” sectionon page 6-17.You can use this simple symmetrical VoIP dial peer technique to join two Cisco Unified CME systemseven within a single site to increase the total phone count supported beyond the capacity of a singleCisco Unified CME system. The downside of doing this is that it does not give you a truly monolithicsystem from a configuration, inter-Cisco Unified CME feature transparency, and management point ofview. This arrangement requires you to administer the two systems separately, which may be acceptableif the two systems are split between naturally different and separate sections of your company (forexample, administration and manufacturing).This arrangement also limits the phone features you can use across the two systems. You can operatesimple features such as call transfer and forwarding, and you can share a single voice mail devicebetween systems, including inter-Cisco Unified CME distribution of message waiting indication (MWI).However, Cisco Unified CME does not support more advanced features, such as shared line and callpickup, across the H.323 or SIP interconnection.Cisco Unified CallManager Express Solution Reference Network Design Guide6-6OL-10621-01
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 NetworkOne final point about this arrangement is that you can optionally choose to provide a physical PSTNconnection on just one Cisco Unified CME system and have that Cisco Unified CME system also act asa VoIP PSTN gateway for the second Cisco Unified CME system.Although this discussion considers H.323 and SIP “long-distance” protocols, the use of these protocolsis not related to physical distance. You can use H.323 and SIP to link systems 1000 feet apart the sameas you would link systems 1000 miles apart. This ability is one of the key advantages of VoIP technologyover traditional TDM systems. With the appropriate IP infrastructure, you can link systems and usersmore or less independent of the physical distance that separates them. This means that you can give aremotely located Cisco Unified CME system a phone extension number and voice mailbox that appearsto your phone users to belong to their local Cisco Unified CME system (with the aforementionedrestriction on advanced phone feature operation between systems across VoIP). The historicalout-of-area-code restrictions that apply to traditional TDM-based centrex phone systems largely do notapply in the VoIP context.The one caveat in this area is the impact on access to public emergency services. Users dialing foremergency assistance (such as police, fire, or ambulance) should be routed into the PSTN via a PSTNconnection that is local to their physical location. The calling party information provided to the PSTNconnection and emergency services operator for this type of call must display an appropriate phonenumber (and therefore an associated physical location) that is within the emergency services area of thePSTN link being used.A Large Multinode Topology with H.323If you want to connect more than two Cisco Unified CME systems, you can extend the basic approachused to connect two systems and add a third, fourth, or more Cisco Unified CME systems—up to a point.For a low number of systems, such as five or six, it’s usually possible to add VoIP dial peers to yourCisco Unified CME system that indicate the static IP address of the other system to reach. This isespecially true if your dial plan is reasonably well segmented such that you can infer to whichCisco Unified CME system the call should be sent based on the first one or two digits of the dialedextension number. For example, Cisco Unified CME system 1 is given extension numbers 5000 to 5099,Cisco Unified CME system 2 is given extension numbers 5100 to 5199, and so on.Even if your dial plan is not entirely evenly divided, you can still use this approach if you are preparedto build the necessary dial peer-based configuration. At the limit of this method, you can constructsystems in which you create an individual H.323 VoIP dial peer on each Cisco Unified CME system foreach remotely located extension number. You can follow this approach as far as available memory andyour patience in creating and maintaining the configuration allow. As the number of dial peers increases,the post-dial delay increases somewhat, because the Cisco Unified CME system might need to searchthrough a couple hundred dial peers to find the right information. In the very worst case, a network offive Cisco Unified CME systems with 20 extensions each would need 80 VoIP dial peers created (andmaintained) on each system. That is assuming that your extension number distribution is fully randomacross the full set of Cisco Unified CME systems. Troubleshooting such a system in the event ofmisconfiguration is challenging, however.Another drawback of the multiple dial peer configuration is that there is no good way to do calladmission control (CAC) in order to prevent an excessive number of voice calls from trying to use thesame WAN link at the same time. This can be an issue if your expected maximum call volume might begreater than the capacity of your WAN links. See the next section for more on this issue.Cisco IOS software has a built-in CAC mechanism with the call threshold interface command. Thisfeature limits both inbound calls and outbound calls for a specific interface on the Unified CME routeronce a maximum threshold has been exceeded. For example, the following command causes calls fromGigabitEthernet0/0 to be rejected after the number of simultaneous inbound/outbound calls exceeds five.Calls are then allowed once the maximum number of simultaneous calls falls below 3.Cisco Unified CallManager Express Solution Reference Network Design GuideOL-10621-016-7
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 Networkcall threshold interface GigabitEthernet0/0 int-calls low 3 high 5The benefit of this feature is that it does not require gatekeeping and can operate across multipledial-peers. It is not subject to dial-peer maximum connection limitations. The limitation of thismechanism is that the maximum number of simultaneous calls is an aggregate of the total inbound andoutbound calls. You cannot set up different thresholds for outbound or inbound calls using thismechanism.Alternatively, you can use the VoIP Tandem Gateway feature of Cisco Unified CME 3.1 and above. Thisallows you to construct hub-and-spoke or hop-by-hop call routing arrangements. Hub-and-spoke callrouting arrangements are historically common in small-scale voice over Frame Relay (VoFR) and voiceover ATM (VoATM) networks. In these small-scale networks, you might have a single larger “hub” CiscoUnified CME system with approximately 100 users at a primary site, with perhaps five satellite CiscoUnified CME systems, each with 20 users linked on VoIP “spokes” to the primary. In this arrangement,only the central hub site needs VoIP dial peers to be configured to define the location of all network-wideextensions. The spoke satellite sites only need to know to send nonlocal calls to the hub site. The centralhub site can then relay the call to the final spoke site destination.This type of arrangement makes the most sense if the physical (Layer 1 and Layer 2) connectivitytopology of your IP transport network mirrors the same hub-and-spoke arrangement as the dial plan.With this situation, IP packets that flow between different spoke sites inevitably get IP Layer 3 routedvia the central hub site Cisco Unified CME router. The hub-and-spoke dial plan arrangement causes theVoIP calls and voice packets to get routed by the application layer instead, with relatively minor addeddelay, as shown in Figure 6-4.Figure 6-4Multinode Cisco Unified CME Tandem Gateway H.323 Network50015002IPIPTandem GatewayV10.5.1.1Cisco Unified CME 1Cisco Unified CME he following example shows the relevant configurations of the nodes shown in the network. Cisco Unified CME 1dial-peer voice 2345 voipdestination-pattern [2345]0.session target ipv4:10.1.5.1no vad Cisco Unified CME 2dial-peer voice 1345 voipCisco Unified CallManager Express Solution Reference Network Design Guide6-8OL-10621-01
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 Networkdestination-pattern [1345]0.no vadsession target ipv4:10.1.5.1 Tandem Gateway Nodevoice service voipallow-connections h323 to h323dial-peer voice 1000 voipdestination-pattern 10.session target ipv4:10.1.1.1no vaddial-peer voice 2000 voipdestination-pattern 20.session target ipv4:10.1.2.1no vadUsing a single dial peer at the spoke sites to direct calls to the hub site and all far-end spoke sites beyondit also allows you to more easily use CAC per dial peer call-counting mechanism (which you’ll learnmore about in the following section). This is shown in Figure 6-4. You can use regular expressions indial peer destination patterns. For example, if you need a single dial peer that references extensions inmultiple ranges, such as 10xx, 30xx, 40xx, and 50xx (not including 20xx), you can use the followingcommand:destination-pattern [1345]0.The values in square brackets ([ ]) provide a list of alternative values—in this case, 1, 3, 4, and 5. You can alsouse this to encompass a continuous range. For example, you can also write the preceding example as1,3-5:destination-pattern [13-5]0.However, an even better and fundamentally more scalable approach to inter-Cisco Unified CME H.323VoIP call routing is to use an H.323 GK (as you will see in the next section). This is the most practicalapproach to link tens or hundreds of Cisco Unified CME systems.The Role of an H.323 GatekeeperThe primary role of an H.323 gatekeeper is to provide a conversion lookup between a telephone numberand an IP address. This service essentially centralizes the dial plan (all the telephone numbers in thenetwork and how to reach them) in a single place in the network, as opposed to each node needing theconfiguration information to do this. This significantly eases the management of a large network.Gatekeepers also provide other services, depending on the type of gatekeeper used. These services arediscussed in this section: Telephone Address Lookup, page 6-11 Call Admission Control, page 6-11 Billing, page 6-12 Using a Gatekeeper as a Proxy for Additional Services, page 6-12Figure 6-5 shows a sample gatekeeper network.Cisco Unified CallManager Express Solution Reference Network Design GuideOL-10621-016-9
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 NetworkMultinode Cisco Unified CME Gatekeeper Network50015002IPIPCisco Unified CME 3V10.5.1.1Cisco Unified CME 1Cisco Unified CME 001149568Figure 6-5The following example shows the relevant dial peer configurations of the nodes shown in the network:NoteNote the use of session target ras; it is explained in the next section. Cisco Unified CME 1dial-peer voice 2345 voipdestination-pattern [2345]0.session target rasno vad Cisco Unified CME 2dial-peer voice 1345 voipdestination-pattern [1345]0.session target rasno vad Cisco Unified CME 3:dial-peer voice 1234 voipdestination-pattern [1234]0.session target rasno vadThe following example shows a more detailed example from an individual Cisco Unified CME routersetup to interwork with an H.323 gatekeeper connected via the Cisco Unified CME router’s Ethernetinterface. Note that the gk ipaddr comand defines the gatekeeper’s IP address.interface FastEthernet0/0ip address 10.1.1.1 255.255.0.0load-interval 30duplex autospeed autono cdp enableh323-gateway voip interfaceh323-gateway voip id gk ipaddr 10.1.10.1 1719h323-gateway voip h323-id cme1Cisco Unified CallManager Express Solution Reference Network Design Guide6-10OL-10621-01
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 Networkh323-gateway voip tech-prefix 1#h323-gateway voip bind srcaddr 10.1.1.1dial-peer voice 1234 voipdestination-pattern [1-4]0.session target rasno vadTelephone Address LookupThe simplest type of gatekeeper provides only telephone number-to-IP address resolution. When aCisco Unified CME system uses a gatekeeper to help route a call, it sends a message to the gatekeeperto request the IP address that corresponds to a certain specific phone number. As soon asCisco Unified CME gets the correct IP address, it can send an H.323 call setup message for the desiredphone number to the IP address of the remote Cisco Unified CME system (provided by the gatekeeper)that hosts that phone number. Instead of having a VoIP dial peer that points to every Cisco Unified CMEsystem in your network, the Cisco Unified CME has only one dial peer that points to the IP address ofthe H.323 gatekeeper.To reference a gatekeeper from a VoIP dial peer, use ras as the target instead of a specific IP address:session target rasIn most cases, the H.323 gatekeeper gets the appropriate phone number-to-IP address configurationdynamically from the component Cisco Unified CME systems. For each individual phone number thatis configured on a Cisco Unified CME system, the Cisco Unified CME system can send a Registrationmessage to the gatekeeper. The Registration message basically says, “I’m an H.323 gateway-like deviceat IP address x.x.x.x, and I have phone number Y.” The gatekeeper aggregates the information from theH.323 Registration messages from all the Cisco Unified CME gateways (and other H.323 gateways) intoa composite database that contains all the current locations of all the telephone numbers in the network.Call Admission ControlIn addition to providing simple telephone number-to-IP address resolution, a gatekeeper can provide calladmission control (CAC) for your VoIP network. CAC keeps track of the number of simultaneous VoIPcalls present at each H.323 gateway and prevents overloading of the gateway’s WAN links (andsometimes also provides load balancing for PSTN access ports). Without CAC, if too many calls attemptto use the same WAN link at the same time, either calls will fail in uncontrolled ways, or too many voicepackets will try to get sent at the same time, leading to voice quality problems.The Cisco Unified CME can do a limited amount of CAC itself without a gatekeeper, either by limitingthe number of simultaneous calls associated with each dial peer or by using an end-to-end bandwidthreservation protocol called Resource Reservation Protocol (RSVP). However, per-dial peer call countingdoes not work well if you are using more than one dial peer per WAN link, and the RSVP mechanismrequires end-to-end support of the RSVP protocol within your network infrastructure, so thegatekeeper-based CAC approach generally is far superior.You can accomplish CAC with the following two Cisco IOS commands: call threshold commandUsing the call threshold command allows you to limit the number of calls allowed through aparticular interface. This can be done with a single Cisco Unified CME. The following is anexample:call threshold interface GigabitEthernet0/1 int-calls low 5 high 5Cisco Unified CallManager Express Solution Reference Network Design GuideOL-10621-016-11
Chapter 6Connecting Multiple Cisco Unified CallManager Express Systems with VoIPIntegrating Cisco Unified CME in an H.323 Network dial-peer commandUsing the dial-peer command limits number of calls based on the max-conn parameter under thedial-peer command. This constraint to the specified maximum number of calls from given dial peer.A caveat associated with this command is that there are usually multiple dial peers on specificCisco Unified CME and the Cisco Unified CME does not track the number calls across all dial peers.If a Cisco Unified CME does have multiple dial peers, with outbound and inbound calls, thedial-peer command solution will not work. If all inbound and outbound calls are routed through asingle dial peer, this command is an effective option. The following is an example of the applicabledial-peer command:dial-peer voice 10 voipmax-conn 10destination-pattern 9Tsession target rasdtmf-relay h245-alphanumericno vadBillingThe gatekeeper keeps track of the number of active calls based on messages from the gateway indicatingwhen individual calls start and stop. Because the gatekeeper knows the start and stop times and the calledand calling phone numbers, a gatekeeper can provide a centralized point to connect to a billing service(for the VoIP calls).This type of billing typically does not know about calls being made by a Cisco Unified CME systemusing its local PSTN connection. These calls do not involve H.323 VoIP call legs, so the H.323gatekeeper typically does not see them. You can use the Cisc
VoIP calls that are from VoIP Public Switched Telep hone Network (PSTN) gateways or even from other independent external/wholesale VoIP carrier networks. You can link independent VoIP networks together and into your corporate VoIP
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