Can't make inbound calls SIP Trunk to cube.

[tresdeasada]

Hi everyone.

I have the following scheme:

SIP Trunk --> CUBE --> CUCM

I can not make inbound calls calls to the number that the phone company gave us..
I have already configured the CSS and Route Patterns in CUCM.

I think I have an incorrect configuration in the CUBE. I am quite new in this and I mostly configured with the guides and the examples that telco gave me.
Maybe some problem with sip-profiles or dial-peers.
I must say that I can only debug inbound calls when I just put my credentials in sip-ua, after a few calls after this, I do not see anything, as if my call did not even reach the router, I put my credentials back and it works again a couple of times.
The external telephone from which I make the call does not ring, only the call is canceled after a while.

Here is my CUBE configuration and "debug ccsip all" when I make an incoming call

Thanks for reading!

[j.huizinga]

Hi

Enable SIP debugs on the CUBE and see what is happening

use: debug ccsip messages

JH

[sansardanish]

Is your registration user ID and password are accurate? Looks like there is some issue there.

Please have a look at the below

https://community.cisco.com/t5/collaboration-voice-and-video/handle-cube-registration-authentication-like-a-boss/ba-p/3660492 

[Shalid Kurunnan Chalil]

Hi,

checked your configuration on registration and it looks good to me. I can also see the registration is successful. you can still verify it by issuing following command - show sip-ua register status .

the logs are broken, but I managed to find out one full call traces and could see call cancel message coming from ISP. 

GW_ERA(config-sip-ua)# 487 Request Cancelled
Via: SIP/2.0/UDP 206.20.196.19:5060;branch=z9hG4bK_1560761911_1225_1
From: <sip:3334893944@206.20.196.19:5060>;tag=1560761911_C
To: <sip:011525513287349@187.192.182.195:5060>;tag=D922EE3C-10C4

I am guessing 206.20.196.19  is ISP IP address. 

one possible issue maybe they do not accept calls on G729, since you are hardcoded to use G729 on your calls.  you can try to remove g729 annexb-all from sip global configuration, also alter the voice codec class if needed. 

Regards,

Shalid 

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