Cisco IOS Voice gateway call flow Troubleshooting commands

[colbea1]

Hello, I am new to the VOIP support world.

I am looking for Cisco IOS Voice Troubleshooting commands/ steps for validating call delivery (Call is reaching the gateway, or why calls are not being delivered on the circuit). Basically, my team is to determine if calls are failing because of a local carrier or if it is something internal to my company. Any help, tips, shortcuts would be greatly appreciated. I understand there are different IOS and all commands don't work on all. So I am grateful for any information that can be shared with a novice.

Example of what was shared with me so far from a co-worker

1. Enable monitoring (term mon)
2. Turn on ISDN Q931 Debugging (debug isdn q931)

[TONY SMITH]

What sort of service is being provided, the debugs you list are correct for an ISDN line, for example for a SIP trunk you would use "debug ccsip mess".   Either way these will show if the call is reaching your gateway, and you can post up an example if you're not sure what it's telling you.

[colbea1]

Hi Tony,

Thank you for the reply. I understand your question. Please reply letting me know how I find out what service is being run. I do know we have both SiP and ISDN in our environment. Also is that typically the only command to run to find out if it is reaching the gateway? If yes the data that is returned is quite a bit of data, what exactly would I be looking for to:

1.  confirm it doesn't reach the gateway 

2. Confirm it does reach the gateway

[TONY SMITH]

Quick way would be to look at the dial peers, either in summary ("sh dial-peer voice sum") or by looking at the full configuration.  If your gateway has both ISDN and SIP, then enable both debugs.   In terms of what you should see, and incoming ISDN call will show an inbound "SETUP" message, for example ...

May 24 07:03:20.112: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8  callref = 0x0001
        Sending Complete
        Bearer Capability i = 0x9090A3
                Standard = CCITT
                Transfer Capability = 3.1kHz Audio
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98381
                Exclusive, Channel 1
        Progress Ind i = 0x8483 - Origination address is non-ISDN
        Calling Party Number i = 0x1180, 'xxxxxxxxx'
                Plan:ISDN, Type:International
        Called Party Number i = 0x81, '1234567890'
                Plan:ISDN, Type:Unknown

If it's SIP it will be an "INVITE" starting something like this ..

Received:
INVITE sip:1234567890@x.x.x.21:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.1:5060;branch=z9hG4bK5bktah002o7gugcok5f0.1

Note I've obscured numbers and IP addresses in those messages.  So you're looking for either a received SETUP with the number in question appearing as "Called Party Number", or a SIP INVITE with the number in either the INVITE line as I've shown, or maybe in a "To:" header further down.

If you have a working number then try that first and compare the results.