ā11-12-2015 02:53 AM - edited ā03-17-2019 04:53 AM
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?
ā11-12-2015 02:59 AM
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
ā11-12-2015 03:50 AM
ā11-12-2015 03:04 AM
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
ā11-12-2015 03:49 AM
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.
ā11-12-2015 06:07 AM
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"
ā11-12-2015 07:21 AM
ā11-12-2015 03:37 PM
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
ā11-13-2015 06:03 AM
ā11-13-2015 09:09 AM
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.
ā11-13-2015 06:23 AM
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?
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