Hello,
Greetings to everyone. I'm having an issue with a new SIP Trunk to CUBE. What is happening the SP is only set to receive invites/etc from the class A address. Which is 4.35.201.X. Right now it is sending the inside address on the invite.
INVITE sip:5122398XXX@4.55.40.XXX:5060 SIP/2.0
Via: SIP/2.0/UDP 10.53.2.215:5060;branch=z9hG4bK1124ED
Remote-Party-ID: "Name" <sip:5124187XXX@10.53.2.215>;party=calling;screen=yes;privacy=off
From: sip:5124187877@10.53.85.15;tag=42A98F88-F50
To: <sip:5122398611@4.55.40.XXX>
Which won't be accepted. I tried binding control and media to the outside interface but I don't even so the call hit the CUBE. Because CUCM can't talk to that address. I believe I could send the from the outside interface if I issued the 'voice-class sip bind control source-interface gig0/1' command on the outbound dial-peer. But we are using version 12.4(24)T6 which doesn't offer that command on the dial-peer. And I don't see anything similar in this version. Any ideas how I could hide the inside interface with this and send the outside address?
voice rtp send-recv
!
voice service voip
allow-connections sip to sip
fax protocol cisco
sip
rel1xx disable
header-passing
early-offer forced
midcall-signaling passthru
sip-profiles 100
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
voice class sip-profiles 100
response ANY sdp-header Audio-Attribute modify "sendonly" "sendrecv"
request ANY sdp-header Audio-Attribute modify "sendonly" "sendrecv"
!
!
interface GigabitEthernet0/0
ip address 10.53.2.XXX 255.255.252.0
no ip route-cache
duplex auto
speed auto
no mop enabled
!
interface GigabitEthernet0/1
description Ckt Id: AUSTTXUH###.
ip address 4.35.XXX.XXX 255.255.255.252
no ip route-cache
duplex auto
speed auto
!
dial-peer voice 100 voip
description 10-digit calls to L3
destination-pattern [2-9].........$
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target ipv4:4.55.40.227
dtmf-relay rtp-nte
!
dial-peer voice 101 voip
description Incoming dialpeer and 1+10 digits to Level 3
destination-pattern ^1[2-9]..[2-9]......$
voice-class codec 1
voice-class sip early-offer forced
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay rtp-nte
sip-ua
retry invite 2
retry bye 2
retry cancel 2
retry register 10
timers expires 60000
sip-server ipv4:4.55.40.XXX
reason-header override
