Here is a link to the h.323 configuration guide.  There is a lot of good info here on h.323.  http://www.cisco.com/en/US/docs/ios/12_3/vvf_c/cisco_ios_h323_configuration_guide/old_archives_h323/323confg.html Here are the highligts of the last h.323/sip trunk implmentation i did.  voice service voip dtmf-interworking rtp-nte allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none h323 sip   early-offer forced   midcall-signaling passthru ! voice class codec 1 codec preference 1 g729r8 codec preference 2 g711ulaw ! voice class h323 1   h225 timeout tcp establish 3   h225 timeout setup 3 ! ! ! (your translation rules will vary depending your needs) voice translation-rule 9 rule 2 /\(^.*\)/ /91\1/ ! voice translation-rule 11 rule 1 /^[1-7]...$/ /5555555555/ ! voice translation-rule 91 rule 1 /^91\(..........\)/ /\1/ rule 2 /^9\(..........\)/ /\1/ rule 3 /^9011\(.*\)/ /011\1/ rule 4 /^9911/ /911/ ! ! voice translation-profile DIGITADD-9 translate calling 9 ! voice translation-profile DIGITSTRIP-9 translate called 91 translate redirect-called 91 (bind h.323 to an interface) interface GigabitEthernet0/0.20 description SIP to CM H323 interface encapsulation dot1Q 20 ip address 172.20.1.3 255.255.255.0 no ip redirects no ip proxy-arp ip nat inside ip virtual-reassembly h323-gateway voip interface h323-gateway voip bind srcaddr 172.20.1.3 (sccp stuff for DSP resources if you are using them) sccp local GigabitEthernet0/0.20 sccp ccm IP OF CUCM identifier 1 priority 1 version 7.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 12 register MTP_DC associate profile 10 register XCODE_DC associate profile 11 register CONF_DC ! dspfarm profile 10 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! dspfarm profile 11 conference description Conference Bridge codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 codec g729br8 maximum sessions 4 associate application SCCP ! dspfarm profile 12 mtp codec g711ulaw maximum sessions software 100 associate application SCCP ! dial-peer voice 10 voip description OUTBOUND Voice SIP calls to VzB translation-profile outgoing DIGITSTRIP-9 destination-pattern 9T session protocol sipv2 session target sip-server voice-class codec 1 voice-class sip early-offer forced dtmf-relay rtp-nte digit-drop no vad ! dial-peer voice 20 voip description INBOUND Voice SIP calls from VzB session protocol sipv2 session target sip-server incoming called-number . voice-class codec 1 dtmf-relay rtp-nte digit-drop no vad ! dial-peer voice 21 voip description to CM from Gateway translation-profile outgoing DIGITADD-9 destination-pattern session protocol sipv2 session target ipv4:ip address of cm incoming called-number . voice-class codec 1 dtmf-relay rtp-nte digit-drop no vad ! dial-peer voice 11 voip description voip dial peer to Verizon translation-profile outgoing DIGITSTRIP-9 destination-pattern 9911 session protocol sipv2 session target sip-server voice-class codec 1 dtmf-relay rtp-nte digit-drop no vad ! dial-peer voice 911 voip description voip dial peer to Verizon destination-pattern 911 session protocol sipv2 session target sip-server voice-class codec 1 dtmf-relay rtp-nte digit-drop no vad ! dial-peer voice 111 voip description voip dial peer to Verizon international translation-profile outgoing DIGITSTRIP-9 destination-pattern 9011T session protocol sipv2 session target sip-server voice-class codec 1 dtmf-relay rtp-nte digit-drop no vad ! ! (This is the SIP information you get from your provider regarding the registar and server they provide you.) sip-ua retry invite 2 retry bye 2 retry cancel 2 retry options 1 registrar ipv4:172.xxx.xxx.164 expires 3600 sip-server ipv4:172.xxx.xxx.49:5091 g729-annexb override I would always do qos.  You could probley get away with using Cisco's auto qos feature unless you have other traffic to account for.  on the switch ports: auto qos voip  cisco-phone On the trunk ports: auto qos voip trust Here is some info on auto qos: http://www.cisco.com/en/US/docs/switches/lan/catalyst6500/ios/12.2SX/configuration/guide/auto_qos.html I hope this helps.  James ... View more

I dont think there is a hard number.  According to the CUCM data sheet CUCM can handle a maximum of 100,000 busy-hour call completions (BHCCs) per Cisco Unified Communications Manager server and 250,000 per server cluster (configuration dependent) For SIP trunks as you mentioned bandwith is one of the factors along with CPU usage, available memory, weather or not  this SIP intergration requires MTP, what codec is being used, transcoding, etc.  Is the SIP trunk terminated on the CUCM or are you using a voice gateway with CUBE?  You have more controll if your using  a VG and CUBE.  For example you can use the max-conn under a dial peer to specify the max number of connections. Or you can use the call threshold global commands to limit calls based on resources available. ... View more