Hey,
I am working on an internal project. Currently I have an ITSP that I am registered with on my CUBE. I have four dial peers configured, some translation rules configured, all listed below.
My internal interface is: Gi0/1.15
My external interface is: Gi0/0
The CUBE router is a Cisco 2911, code 15.7
The CUCM publisher is at IP 192.168.15.20
I have some DIDs with the ITSP, but I am currently just working on getting one internal phone working with inbound/outbound. Currently inbound dialing is working correctly but with one way audio, I will fix that later. I'm sure it has to do with the VoIP phone being on a DMVPN spoke router remotely. Outbound dialing I am getting your call cannot be completed as dialed.
On the CUCM side, it's very simple. I have a phone with extension 7575 on it that CUBE is translating calls to. For my route pattern, I have it set to strip the predot, so CUCM is sending 1xxxxxxxxxx to the CUBE. The CUBE should then send calls to the ITSP with the 1. It appears I am matching dial-peer 3 when going outbound from the phone. There is a CSS and partition that I am giving access to everything.
Any ideas where I went wrong with my dial peers?
Here's my relevant config:
voice translation-rule 1
rule 1 /.*/ /***DID***/
!
voice translation-rule 2
rule 1 /***DID***/ /7575/
!
voice translation-profile INCOMING_FROM_PSTN
translate called 2
!
voice translation-profile outgoing_cid
translate calling 1
!
dial-peer voice 1 voip
description *** Outbound dial-peer to ITSP ***
translation-profile outgoing outgoing_cid
destination-pattern 1[2-9]..[2-9]......T
session protocol sipv2
session target sip-server
voice-class sip early-offer forced
voice-class sip bind control source-interface Gi0/0
voice-class sip bind media source-interface Gi0/0
dtmf-relay rtp-nte h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 2 voip
description *** Incoming Dial-Peer from ITSP ***
translation-profile incoming INCOMING_FROM_PSTN
session protocol sipv2
session target ipv4:192.168.15.20
incoming called-number ***DID***
voice-class sip early-offer forced
voice-class sip bind control source-interface Gi0/1.15
voice-class sip bind media source-interface Gi0/1.15
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 3 voip
description *** TRUNK FROM CUCM ***
session protocol sipv2
session target sip-server
incoming called-number .
voice-class sip bind control source-interface Gi0/1.15
voice-class sip bind media source-interface Gi0/1.15
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 4 voip
description *** TRUNK TO CUCM ***
destination-pattern ^7575$
session protocol sipv2
session target ipv4:192.168.15.20
voice-class sip bind control source-interface Gi0/1.15
voice-class sip bind media source-interface Gi0/1.15
dtmf-relay rtp-nte
codec g711ulaw
no vad
Here's some debugs:
Edit:
Ran one more debug
Looks like I'm not matching a dial peer outbound? Where am I missing?
Route Pattern screenshot:
Thanks!
Let me know if you need specific debugs, screencaps, etc.
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