SIP Trunk:

========
CallManager versions before 5.x can only have a SIP trunk to a gateway or other
servers. CallManager 5.x and above also acts as a SIP B2BUA and allows SIP phone
s to register to it
To change the transport protocol (UDP is the default), use the session transport
.
A basic SIP gateway configuration consists of simply adding one line under a VoI
P dial peer configuration: session protocol sipv2.
sample config:
--------------
SIP-GW(config)#dial-peer voice 3401 voip
SIP-GW(config-dial-peer)#session target ipv4:10.6.2.1
SIP-GW(config-dial-peer)#session protocol sipv2
SIP-GW(config-dial-peer)#session transport tcp
!
SIP-GW(config)#dial-peer voice 4404 voip
SIP-GW(config-dial-peer)#session target ipv4:10.7.1.1
SIP-GW(config-dial-peer)#session protocol sipv2
SIP-GW(config-dial-peer)#voice-class sip transport switch udp tcp
SIP-GW(config-dial-peer)#destination-pattern 4404...
SIP-GW(config)#voice service voip
SIP-GW(conf-voi-serv)#redirect ip2ip
SIP-GW(conf-voi-serv)#sip
SIP-GW(conf-serv-sip)#bind control source-interface lo0
SIP-GW(conf-serv-sip)#registrar server expires max 1500 min 500
SIP USER AGENT configuration:
------------------------------
SIP-GW(config)#sip-ua
SIP-GW(config-sip-ua)#registrar ipv4:10.30.25.250 tcp
SIP-GW(config-sip-ua)#registrar ipv4:10.30.25.251 tcp secondary
SIP-GW(config-sip-ua)#sip-server ipv4:10.30.25.252
SIP-GW(config-sip-ua)#max-forwards 10
SIP-GW(config-sip-ua)#no transport udp
several configurations are availoable in sip-ua:
timers, nat, calling-info, transport {udp | tcp}, sip-server (this specifies the
sip server), registrar
Verification:
------------
show sip-ua status
show sip-ua register status
The voice service configuration mode is used for voice-related commands that aff
ect the entire gateway--->voice service voip
Configuration of SIP trunk in call Manager:
Note that a media termination point (MTP) is required-->This is to translate DTM
F signals between SIP and SCCP phones. For more information
CallManager 5.x can register SIP phones, in addition to SIP trunks. Trunks can p
oint to other Cisco CallManager clusters, also.
Allow h323 to sip calls:
----------------------
H323-SIP-GW(config)#voice service voip
H323-SIP-GW (conf-voi-serv)# allow-connections h323 to h323
H323-SIP-GW (conf-voi-serv)# allow-connections h323 to sip
H323-SIP-GW (conf-voi-serv)# allow-connections sip to h323
H323-SIP-GW (conf-voi-serv)# allow-connections sip to sip