ISR Debug Recommendation Cisco Telephony

Posted on 2013-12-24
Last Modified: 2013-12-25
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.
Question by:amigan_99
Welcome to Experts Exchange

Add your voice to the tech community where 5M+ people just like you are talking about what matters.

  • Help others & share knowledge
  • Earn cash & points
  • Learn & ask questions
LVL 63

Accepted Solution

btan earned 500 total points
ID: 39739026
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

Author Closing Comment

ID: 39739234
This is really an outstanding answer.  Wish I could rate Excellent Plus!  Thank you.

Featured Post

Save the day with this special offer from ATEN!

Save 30% on the CV211 using promo code EXPERTS30 now through April 30th. The ATEN CV211 connects a laptop directly to any server allowing you instant access to perform data maintenance and local operations, for quick troubleshooting, updating, service and repair.

Question has a verified solution.

If you are experiencing a similar issue, please ask a related question

Suggested Solutions

Hi there, This article summarizes what you need if you are going to set up your home or small business Network Attached Storage (NAS) to be accessible from the internet. Of course there are configuration differences based on your NAS or router ma…
Problem Description:   Couple of months ago we upgraded the ADSL line at our branch office from Home to Business line. The purpose of transforming the service to have static public IP’s. We were in need for public IP’s to publish our web resour…
After creating this article (, I decided to make a video (no audio) to show you how to configure the routers and run some trace routes and pings between the 7 sites…
After creating this article (, I decided to make a video (no audio) to show you how to configure the routers and run some trace routes and pings between the 7 sites…

726 members asked questions and received personalized solutions in the past 7 days.

Join the community of 500,000 technology professionals and ask your questions.

Join & Ask a Question