Why aren't my Dial peers working?

[richard.priest]

I'm trying to setup my own CUCM homelab, I've got most it working at a basic level, incoming calls work perfectly. However outgoing calls fail

I have a SIP trunk between CUCM and a CSR1000v which is being used as a CUBE. I then have a SIP provider for the external access.

As I understand it I need x2 Dial peers for each leg, one in, and one out. (from the perspective of the CUBE).

So for external calls into my CUCM setup I have the following x2 dial-peers

dial-peer voice 200 voip
description **Incoming calls CUBE to CUCM ***
destination-pattern .T
session protocol sipv2
session target ipv4:192.168.37.200
voice-class codec 1
voice-class sip early-offer forced
dtmf-relay rtp-nte sip-notify
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 20 voip
description ***Inbound WAN (SIPGATE)***
translation-profile incoming SIP-INBOUND
session protocol sipv2
session target ipv4:192.168.37.200
incoming called-number .
voice-class codec 1
voice-class sip privacy-policy passthru
voice-class sip early-offer forced
voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay rtp-nte
dtmf-interworking rtp-nte
ip qos dscp cs3 signaling
no vad

and the following two for outbound

!
dial-peer voice 10 voip
description ***Outbound to SIPGATE***
translation-profile incoming 6
translation-profile outgoing 6
destination-pattern .T
session protocol sipv2
session target dns:sipconnect.sipgate.co.uk
session transport udp
voice-class codec 1
voice-class sip privacy-policy passthru
voice-class sip options-ping 60
voice-class sip early-offer forced
voice-class sip profiles 101
voice-class sip bind control source-interface GigabitEthernet1
voice-class sip bind media source-interface GigabitEthernet1
dtmf-relay sip-notify rtp-nte
clid network-number 2723959
no vad
!
dial-peer voice 100 voip
description *** Outgoing calls CUCM to CUBE***
translation-profile incoming 6
translation-profile outgoing 6
session protocol sipv2
session target dns:sipconnect.sipgate.co.uk
incoming called-number 9.T
voice-class codec 1
dtmf-relay rtp-nte sip-notify
no vad

Note that the translation profiles don't seem to be working, I had to set the calling party mask in CUCM / Route pattern to get the logs to show a full dial code.

I've attached a log of debug ccsip messages. You can see I get a 407 proxy authentication required, and an ACK afterwards so I assume that's OK?

But then I get a 403 forbidden. I can't work out why I'm getting that though?

In addition to this outgoing calls don't seem to be using dial-peer 100 as I expected them to. They're using dial-peers 10 and 200? I've also attached a debug voice dial-peer all of an outbound call

Any tips and pointers are much appreciated.

[TONY SMITH]

It's difficult to see all the details, for example you don't show the translation profiles or rules. 

Your dial peer 100, is that intended to match the inbound leg from CUCM -> CUBE?  You have it configured to match a called number of 9.T, but in your debugs both called and calling numbers start 0.   

[richard.priest]

Yeah sorry I should've included my confg.

I've messed around some more and I think I'm closer, but it's still not working for outbound calls.

I've attached my config and new debugs.

[TONY SMITH]

Is the device registering successfully with Sipgate (sh sip-ua register status)?   I have some Wireshark traces for Gigaset to Sipgate, and one difference I notice is that I was registering to "sipgate.co.uk" rather than "sipconnect.sipgate.co.uk".

Other than that the call sequence in my trace looks the same, initial INVITE receives 407, then the following INVITE includes the proxy authorization header. 

[richard.priest]

Thanks Tony,

Registration is showing successful (I'd assume that inbound wouldn't work either if registration wasn't working?

All the documentation I have from SIPGate states that I should use sipconnect.sipgate.co.uk. If I try to use just sipgate then the registration fails