Need help with logging on 1720s

[bigblock454]

Hi All, I need some assistance with setting up logging on several 1720s, running from 12.0 to 12.5. I need this to apply to all tcp/ip traffic and especially traffic related to VoIP, I just dont' know what to set or how.

Thanks, Dawn

 

[Minue]

Hello
It's not clear what you want to actually do.Would it be traditional logging for router notifications,warnings,etc.like interface went down or up,protocol changes.Or you want to monitor traffic flows,on the network.There's a bit of difference between the two.the First is simple just download a syslog Server from the Internet or you can purchase one,Instruction to install are quit clear.
As for monitoring tcp/ip you can use Netflow,IP accounting.
Regards

 

[bigblock454]

Hi Minue, I'm not sure what they want, really. We installed a VoIP (Avaya IPO) in March of this year, and have had extensive problems. I have spent literally hundreds of hours trying to resolve why we get dropped calls on a regular basis, and our vendor has been very little help. We have tested everything every way you can imagine, replaced equipment, rewired, etc. until I am so sick of it I could scream. I had folks on the IPO forum working with me literally for weeks on end earlier this year, and they are all great, but the problem persists. I am not a total novice, I've been in computing for 13 years and worked network engineering for Sprint for 2, although that was strictly data on the national internal network. BurstBees earlier this year assisted me in setting up a QoS policy for VoIP on my 1720s.

My (IPO) vendor wants to blame my network, although I have not once seen anything that correlates or shows causation with the problem. I have used ethereal, solar winds, and PRTG traffic monitor,and others to try and track any problems that might exist on the network, all to no avail. Even when I can see packet loss, it never corresponds with the drops. So just exactly WHAT they want, I have no idea at this point. If I sound frustrated, well, I am...I will take a look at the programs you suggested too, thanks.

 

[brianinms]

Most reliable companies test the network and verify its interoperability for VOIP before the product is even sold.

 

[bigblock454]

brianinms, yes, you would think so...but no...it has been a HUGE sore point, and I've almost resigned several times over this whole debacle...But since I'm stuck with what I've got, well...Someone on another thread with a different problem mntioned something about the signalling, anyone think that might have an impact? I have 4 sites, 3 are connected to the main by PTP T1, 2 of 3 run AMI/SF, the other B8ZS/ESF (that's the one with the most problems). Probably the wrong place to ask, but hey, I'll take whatever I can get.

Thanks, Dawn

 

[brianinms]

I doubt signaling of the T1 would have any effect on voice, however quality of service such as bandwidth, latency, and jitter all play a role in voice quality.

So I take it the dropped calls are between corporate and the branches?

 

[bigblock454]

brianinms, generally, no. The drops can be anything, but the vast majority are incoming calls. What makes it stranger is the fact that they tend to be in clusters of the same caller. So caller A calls in and is dropped, then calls back and is dropped, and so on. Clusters range from 2-6 drops within approximately a 10min-30min window. Other callers in the same window are not dropped. This can occur with any kind of line, wired or wireless, and are from various LATAs. A signifigant amount are transfers from one specific location to another, but none of these things are absolute, all have exceptions, and all locations have had drops. The one constant is the error received on the IPO Monitor, it is always the same, here is an example:

440270046mS ERR: EXCEPTION ON MEDIA CONNECTION --- Error from protocol entity! MSDSE --- local timer expired ...
440270046mS CMLOGGING: CALL:2007/12/1210:54,00:01:29,013,5733785309,I,536,536,,,,0,,""n/a,0
440270047mS CD: CALL: 40.411.1 State=2 Cut=3 Music=0.0 Aend="Line 40" (0.0) Bend="" [Troy Sage(511)] (10.3) CalledNum=511 (PartsGroup21mm) CallingNum=5733785309 () Internal=0 Time=103049 AState=2
440270047mS CD: CALL: 40.411.1 Deleted
440270050mS CMLineTx: v=40
CMReleaseComp
Line: type=IPLine 40 Call: lid=40 id=411 in=1
Cause=2, No route to specific transit network/(5ESS)Calling party off hold
440270051mS CMCallEvt: 40.411.1 -1 H323TrunkEP: StateChange: END=X CMCSConnected->CMCSDelete
440270053mS CMCallEvt: 0.4243.0 -1 Troy Sage.0: StateChange: END=X CMCSConnected->CMCSCompleted
440270054mS CMExtnTx: v=511, p1=0
CMReleaseComp
Line: type=DigitalExtn 4 Call: lid=0 id=4243 in=0
Cause=2, No route to specific transit network/(5ESS)Calling party off hold
Timed: 12/12/07 10:56
440270055mS CMCallEvt: 0.4243.0 -1 Troy Sage.-1: StateChange: END=X CMCSCompleted->CMCSDelete
440270056mS CMCallEvt: END CALL:901

 

[bigblock454]

P.S. Everything I know about Avaya IPO I have learned here on tek-tips, btw. I knew nothing about phone systems and VoIP when this was thrown at me, sorry that it's gotten off topic, but I truly appreciate any ideas.

 

[brianinms]

I am a cisco guy myself so I am no help with Avaya, but general troubleshooting principles still hold true. In all honestly I would call Avaya and find a different partner.

However, back to your original request ... that statement is too vague. You would be better off configuring IP SLA and monitoring your conditions than trying to read through logs afterwards. Additionally paying a cisco partner to come in and perform a review and assessment of your network would be money well spent.

http://www.cisco.com/application/pd...ps6350/c2001/ccmigration_09186a0080789b77.pdf

 

[bigblock454]

brianinms, thanks. I will read over that document and give it a shot.

As far as another business partner, I've tried, but the boss won't go for it <sigh>. Would the Cisco website be a good resource to find a competent Cisco partner?

Thanks again, Dawn

 

[Minue]

Hello
Please post your Qos config and some "show policy-map interface"
The first thing to see if the voice is getting enough bandwith.
Regards

 

[brianinms]

Yes, you can browse to

http://cisco.com/pcgi-bin/cpn/cpn_pub_bassrch.pl

to find a partner in your area. I would look for a local Gold or Silver partner.

 

[bigblock454]

Hi Minue, and thanks. Here is my Qos:
class-map match-all VoIP
match ip dscp ef
match ip dscp af41
!
!
policy-map voipQos
class VoIP
priority percent 50
class class-default
fair-queue
random-detect dscp-based

And the sho policy-map interface:
Serial0

Service-policy output: voipQos

Class-map: VoIP (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: ip dscp ef (46)
Match: ip dscp af41 (34)
Queueing
Strict Priority
Output Queue: Conversation 264
Bandwidth 50 (%)
Bandwidth 772 (kbps) Burst 19300 (Bytes)
(pkts matched/bytes matched) 0/0
(total drops/bytes drops) 0/0

Class-map: class-default (match-any)
17954869 packets, 3638790453 bytes
5 minute offered rate 8000 bps, drop rate 0 bps
Match: any
Queueing
Flow Based Fair Queueing
Maximum Number of Hashed Queues 256
(total queued/total drops/no-buffer drops) 0/244/0
exponential weight: 9

dscp Transmitted Random drop Tail drop Minimum Maximum Mark
pkts/bytes pkts/bytes pkts/bytes thresh thresh prob
af11 0/0 0/0 0/0 32 40 1/10
af12 0/0 0/0 0/0 28 40 1/10
af13 0/0 0/0 0/0 24 40 1/10
af21 0/0 0/0 0/0 32 40 1/10
af22 0/0 0/0 0/0 28 40 1/10
af23 0/0 0/0 0/0 24 40 1/10
af31 0/0 0/0 0/0 32 40 1/10
af32 0/0 0/0 0/0 28 40 1/10
af33 0/0 0/0 0/0 24 40 1/10
af41 52673/4140358 0/0 0/0 32 40 1/10
af42 0/0 0/0 0/0 28 40 1/10
af43 0/0 0/0 0/0 24 40 1/10
cs1 0/0 0/0 0/0 22 40 1/10
cs2 0/0 0/0 0/0 24 40 1/10
cs3 0/0 0/0 0/0 26 40 1/10
cs4 0/0 0/0 0/0 28 40 1/10
cs5 0/0 0/0 0/0 30 40 1/10
cs6 0/0 0/0 0/0 32 40 1/10
cs7 0/0 0/0 0/0 34 40 1/10
ef 5407111/1094358276 0/0 0/0 36 40 1/10
rsvp 0/0 0/0 0/0 36 40 1/10
default 12495315/2540012229 244/343657 0/0 20 40 1/10

 

[Minue]

Hello
The voice traffic isn't being match in the priority queue.I am seeing voice traffic in "class class-default" with fair-queue.Try taking out this command "random-detect dscp-based"You are having drops and the voice could be in the middle.
Regards

 

[Minue]

Hello
Just took a better look at your config,and I think your problem is the class-map.You need a match-any not match-all.With the "all" the policy will only apply if the router see's EF and af41 packets together.This is a bit
tricky.
Try the "class-map match-any VoIP"in that way the policy will match EF or af41 when it see's it.
Goof luck

 

[kwing112000]

Bigblock,
Have you had a network assesment yet. We run them all the time and what we do is set up a PC on each end of the link and flood the connection with VOIP packets. this will test the QOS policies and verify they are doing what they need. Now maybe i am wrong but the QOS only really kicks in when the interface if full of traffic. we are an Avaya BP and we have some good network engineers. It may finally be time to ditch your vender and find someone who knows what is going on. Let me know and i can have some one call you if you are interested. I feel bad i have seen you struggling with this for a long time on here.

 

[kurthansen]

Dawn,

Do you have any switches between the IPO and the routers? Could you post the config of those also?

 

[Minue]

Hello
Did you resolved your problem?WE would be happy to know.
Regards

 

[auEric]

When do the calls drop? At the begining or part way through call?