Confused by basic SIP Trunk configuration.

Question

I've went through a few basic SIP trunk configurations and Youtube videos the last couple days but can't figure out what I'm doing wrong.
I've set up H323 and MGCP no problem, but I can't figure out the SIP trunk set up. I'm guessing there are some concepts I'm not understanding yet.
I've got a CUCM lab set up. A 2851 PSTN Simulator, 2851 H323 Gateway at the Main site with a 9.0 CUCM setup in that site and a Branch site that I'm trying to set up as a SIP trunk to connect two phones.
CUCM is on the 192.168.5.x/24 subnet. 172.16.0.x/24 is the subnet connecting the serial(internet) cable between the two gateways in which I'm trying to establish the trunk between.
The Branch phones are still registering with the CUCM at the main site. The Route Pattern is looking to the Branch Route List which has the SIP Trunk listed. I'm just getting a fast busy when trying to place a call from the branch site to the main site.
The most frustrating thing I'm not understanding, is that the debug ccsip and call debugs on my SIP Branch gateway shows absolutely nothing.  I've tried registering the branch phones with the SIP Trunk, but stopped when I figured that shouldn't be necessary.
If someone can make some sense of this, I'd truly appreciate it!

Answer 1

Hello Aditya and thanks for the consideration!
I do have a direct IP connection, but I want to set up a SIP trunk and use it just to know how to do it before I do it in production. 
I did end up deleting the phones from CUCM so they can register with the 2851 CME that I'm setting up as a SIP trunk. So it is registering there, and I set the allow connections and bind sip commands.
I am now getting Debugs and calls from the SIP Trunk router going to CUCM, but the error message is No Codec, and I Get the fast busy after the call rings on the CUCM Main Site side. So looks like the negotiation is failing. Here is my CLI for the SIP Trunk now after the changes have been made and phones registered to the SIP Branch site as well as the Debug when I tried to place a call to extension "5000":
Note: I did try to change the codecs in the dial-peers to g729r8 instead of 711 and same fast busy after answering.
==============================================
Branch_SIP#show run
Building configuration...
Current configuration : 3529 bytes
! Last configuration change at 03:15:11 UTC Thu Apr 2 2015
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Branch_SIP
boot-start-marker
boot-end-marker
! card type command needed for slot/vwic-slot 0/2
enable secret 5 $1$hOXF$gvfmWW1ZIQE0mAMVg.u1c/
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 10.0.10.1 10.0.10.10
ip dhcp excluded-address 10.0.30.1 10.0.30.10
ip dhcp pool Data
 network 10.0.10.0 255.255.255.0
 default-router 10.0.10.254
 option 150 ip 192.168.5.250
 dns-server 192.168.5.200
ip dhcp pool Voice
 network 10.0.30.0 255.255.255.0
 default-router 10.0.30.254
 dns-server 192.168.5.200
 option 150 ip 172.16.0.1
ip dhcp pool data
 option 150 ip 172.16.0.2
no ipv6 cef
multilink bundle-name authenticated
voice service voip
 allow-connections sip to sip
 sip
  bind media source-interface Loopback1
voice-card 0
crypto pki token default removal timeout 0
license udi pid CISCO2851 sn FTX1031A2FM
redundancy
interface Loopback1
 ip address 2.2.2.2 255.255.255.255
interface GigabitEthernet0/0
 no ip address
 duplex auto
 speed auto
interface GigabitEthernet0/0.10
 encapsulation dot1Q 10
 ip address 10.0.10.254 255.255.255.0
interface GigabitEthernet0/0.30
 encapsulation dot1Q 30
 ip address 10.0.30.254 255.255.255.0
interface GigabitEthernet0/1
 no ip address
 shutdown
 duplex auto
 speed auto
interface Serial0/3/0
 no ip address
 shutdown
 clock rate 2000000
interface Serial0/3/1
 ip address 172.16.0.1 255.255.255.0
 clock rate 250000
interface Internal-Service-Module0/0
 no ip address
 shutdown
 !Application: CUE Running on AIM2
 hold-queue 512 out
router eigrp 1
 network 0.0.0.0
 network 2.2.2.2 0.0.0.0
 network 10.0.0.0
 network 10.0.10.0 0.0.0.255
 network 10.0.30.0 0.0.0.255
 network 172.16.0.0
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 0.0.0.0 0.0.0.0 172.16.0.2
tftp-server flash:term45.default.loads
tftp-server flash:jar45sccp.8-5-3TH1-6.sbn
tftp-server flash:cnu45.8-5-3TH1-6.sbn
tftp-server flash:apps45.8-5-3TH1-6.sbn
tftp-server flash:dsp45.8-5-3TH1-6.sbn
tftp-server flash:cvm45sccp.8-5-3TH1-6.sbn
control-plane
voice-port 0/0/0
voice-port 0/0/1
mgcp profile default
dial-peer voice 1 voip
 description **Incoming Call from SIP Trunk**
 session protocol sipv2
 session target sip-server
 codec g711ulaw
dial-peer voice 2 voip
 description **Outgoing Call to SIP Trunk**
 destination-pattern 5...
 session protocol sipv2
 session target sip-server
 codec g711ulaw
sip-ua
 sip-server ipv4:192.168.5.250
telephony-service
 codec g711ulaw
 max-ephones 24
 max-dn 48
 ip source-address 172.16.0.1 port 2000
 system message SIP Branch Site
 cnf-file location flash:
 load 7960-7940 P00308010200.bin
 max-conferences 8 gain -6
 transfer-system full-consult
ephone-dn  1
 number 4008
ephone-dn  2
 number 4005
ephone  1
 device-security-mode none
 mac-address 001D.A21A.2065
 button  1:1
line con 0
 exec-timeout 0 0
line aux 0
line 194
 no activation-character
 no exec
 transport preferred none
 transport input all
 transport output lat pad telnet rlogin lapb-ta mop udptn v120 ssh
 stopbits 1
 speed 115200
line vty 0 4
 password cisco
 login
 transport input all
line vty 5 15
 password cisco
 login
 transport input all
scheduler allocate 20000 1000
end
Branch_SIP#show debug
TFTP:
  TFTP Event debugging is on
CCSIP SPI: SIP Call Statistics tracing is enabled       (filter is OFF)
Branch_SIP#
*Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x4B6C5C28
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 4008
Called Number            : 5005
Source IP Address (Sig  ): 172.16.0.1
Destn SIP Req Addr:Port  : 192.168.5.250:5060
Destn SIP Resp Addr:Port : 192.168.5.250:5060
Destination Name         : 192.168.5.250
*Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : No Codec
Negotiated Codec Bytes   : 0
Nego. Codec payload      : 255 (tx), 255 (rx)
Negotiated Dtmf-relay    : 0
Dtmf-relay Payload       : 0 (tx), 0 (rx)
Source IP Address (Media): 2.2.2.2
Source IP Port    (Media): 19472
Destn  IP Address (Media):  -
Destn  IP Port    (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
*Apr  2 03:20:32.351: //25/0C496935804A/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 63
Disconnect Cause (SIP)   : 503
Branch_SIP#