ISR Debug Recommendation Cisco Telephony

I have a site where every 10 days or so I start getting complaints that outbound dialing is failing.  Typically the far end never starts ringing.  And its inconsistent.  Some folks are dialing ok but others not.  CPU and memory are fine.  I don't see the router under any kind of DOS attack.  I would like to just syslog some kind of debug to troubleshoot this better at the next occurance.  What level of debug would be recommended?  It uses SIP to communicate with the PSTN provider.  And internally it is a SIP trunk (CUCM sees it as a SIP voice trunk at a particular IP address.  Not an MGCP gateway or other.)  Currently I am running debug ccsip messages and the CPU is staying at a comfortable 5% with a very occasional brief peak to 25% or so.
amigan_99Network EngineerAsked:
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btanConnect With a Mentor Exec ConsultantCommented:
I am not totally savvy in voip but hopefully these augment your current effort with use workflow below so that you can isolate the issue:
1. Call Flow between PBX to Cisco SIP IP Phone—Successful Setup and Disconnect
2. Call flow between Gateway-to-Cisco SIP IP Phone Call—Successful Call Setup and Call Hold
3. Call flow between Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold

Typically the full trace of your test call are recorded using
-debug ccsip messages
-debug voip rtp session named-event

Cisco IOS Voice Troubleshooting and Monitoring -- Cisco SIP Gateway Troubleshooting

Troubleshoot a SIP Call Between Two Endpoints - provides an explanation on the output of the debug ccsip messages command for troubleshooting SIP call failures.

Basic SIP Call Flows & Troubleshooting Commands

DISSECTING A SIP TRACE (as below), include (further down in that article threads discusssion) PSTN-------->ITSP------->CUBE--------------->CUCM---------------->IP PHONE

Finally the call is ended. Now when troubleshooting the direction of call termination is important. In this case we can see that the CUBE receives a BYE, which is the sip method for call termination. However who sent the BYE, is it CUCM or ITSP…The answer is in the Call-ID. As we call can see the CALL-ID is for the leg from the ITSP. So we see that the call was terminated from the ITSP side.

Some problem shared in forum include

Problem was that my incoming dial-peer from CUCM was set to use G729r8 and my outgoing SIP dial-peer was set to use G729br8 ,so the router was trying to transcode ,don`t know how i missed this. ...

You are sending DTMF with PT 97, that is payload type 97. However in your SDP extensions, there are no attribiutes for this Payload type.
ACCDMM-VGW01(config-dial-peer)#rtp payload-type nte 97
ERROR: value 97 in use!
Do the ff: rtp payload-type cisco-codec-fax-ack 98
 then do this again and test rtp payload-type nte 97

on the outbound direction...
The advertised codec to ITSP will be used, beacuse ITSP is the one making decision which codec to use based on advertised codec.So here with voice class codec, the preffered codec in the list will be selected by the ITSP and this will be used for the call regardless of what the region setting is on the phone to the cube gateway.
if the codec is hardcoded..if its set to g711, then g711 response will be obtained from the ITSP and that codec will be used for the call.

So the outbound leg is independent of the region settings beacuse it is the far end that is choosing
what codec to use for the call.
So in both cases the codec selected by the far end will determine what codec is used for the call
Inbound leg, far end is CUCM..region takes effect
outbound leg, far end is ITSP, prefered codec in advertised codec will take effect..Simples!
So if you want to choose a codec for your outbound leg, you have to either set it as preferred in your voice class list or hardcode it such that it is the codec been advertised as the prefferd to your ITSP
amigan_99Network EngineerAuthor Commented:
This is really an outstanding answer.  Wish I could rate Excellent Plus!  Thank you.
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