Dial peer not forwarding calls to CUCM

[MastahBates]

Hi All,

I am having a few problems getting external calls from a SIP circuit reaching my CUCM server.

I have a range of DDIs (01539760560 through to 01539760574) and I will be setting up the extensions on CUCM to be between 5560 - 5574.

I have attached the current running configuration of the voice gateway as well as the output of the "debug voip dialpeer" and I can see that the number being called (01539760560) is being translated by the num-exp to 5560 and that the dial-peer is matching successfully to dial-peer 55 which has the session target as CUCM.

Am I missing something?

[Rajan]

Hi Mastah,

We are indeed hitting the dial-peer for sending the call to CUCM. Could you also provide debug voip ccapi inout to check further.

Also provide the CUCM traces with the call details to see whether the call is hitting the CUCM.

Thanks,

Rajan

[MastahBates]

Hi Rajan, 

Please find the output attached to this post and also the reply to the post above yours.

[Vivek Batra]

Which protocol is intended to use between gateway and CUCM, SIP or H323?

You haven't defined SIP bind interface. You should define globally and under dial-peer preferably.

No codec or voice-class codec is defined under dial-peer. 

Please share the output of debug voice ccapi inout and debug ccsip messages to check further.

- Vivek

[MastahBates]

The gateway on CUCM is configured as a h323 gateway - would this be the root of my problems?

I have added the following commands to dial-peer 55:

voice-class codec 1

voice-class h323 1

dtmf-relay h245-alphanumeric

fax-relay ecm disable

fax rate disable

no vad

Also, find attached a pretty extensive output from the two commands.

[Chris Deren]

You dont have any inbound dial-peers.

I would recommand converting CUCM-GW integration to SIP to stay consistent and ease of troubelshooting. After conversion to sip post "debug ccsip messages"

[MastahBates]

Hi,

I have updated the config; it now shows four call legs (two inbound, two outbound) as well as some sip-ua config too.

I have also attached debug voice ccapi inout and debug ccsip message outputs after applying the updated config.

Thanks

[Ayodeji Okanlawon]

You need to add your voice class codec to dial-peer 2..

dial-p v 2 v

voice-class cod 1

Test again and send 

Debug ccsip messages 

Debug h225 asn1

Debug h245 asn1

Debug voip ccapi inout

[MastahBates]

Hi Ayodeji,

I have attached the updated configuration as well as the output of the debugs you have asked for.

Just out of curiosity so that I can learn myself; what is it you typically look for in these debug outputs?

Thanks.

[Ayodeji Okanlawon]

You need to enable fast start on your h323 gateway on cucm.  Your ITSP is sending INVITE using early offer, hence you have to enable inbound fast start on the gateway. 

Ensure you reset the gateway after enabling this. 

[MastahBates]

This is an excerpt of a debug I have ran:

Nov 13 13:59:45.693: //88/9E400A6C810D/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 88.215.63.9:5060;branch=z9hG4bKfe3e3c6fbe7ebc01c6214db98252e5d6

From: <sip:07718572110@88.215.63.9>;tag=r94y9vij46a

To: <sip:01539760560@88.215.63.9:5060;user=phone>;tag=6270058-153C

Date: Fri, 13 Nov 2015 13:59:42 GMT

Call-ID: 4038388-3656411980-527232@MSX75.gammatelecom.com

CSeq: 1 INVITE

Require: 100rel

RSeq: 989

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

Allow-Events: telephone-event

Contact: <sip:01539760560@172.12.10.253:5060>

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

I would have expected the contact SIP address would be 5560@17.16.210.10 (CUCM IP Address)? Wouldn't any of you?