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Call Manager Express - Small Configuration Issues

Posted on 2010-09-03
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Last Modified: 2012-05-10
Experts,

This isn`t an emergency by any means...  BUT it is very annoying.  :)

I have a 2600xm router configured with CME 3.3.   In this setup, I have one of the voice cards, with some FXO ports, that I receive incoming calls, through the automated attendant.  Everything works great as configured...

Except - sometimes when an external line attempts to connect through the system, but the external POTS phone disconnects before the IP phone user picks up - the IP phone will ring FOREVER - until the IP phone is taken off-hook.

If that makes sense.      Like in this example...   A user in POTS calls the business number after hours.  Nobody inside the network picks up the phone.  But it continues to ring, continually - until users come to work in the morning and off-hook their phones.

Is there any way to limit the number of rings on inbound phone calls from POTS, prior to simply disconnecting them?

Also, when I park a call in one of the slots, I have 5001, 5002, and 5003 as park slots...  When a call is parked there - I get busy signals when I try to pick up the call from any internal phone.

My current voice config is pasted below.

Any ideas would be greatly appreciated.
voice service voip
 allow-connections h323 to h323
 allow-connections h323 to sip
 allow-connections sip to h323
 allow-connections sip to sip
 redirect ip2ip
 sip
  bind control source-interface FastEthernet0/0
  bind media source-interface FastEthernet0/0
  registrar server expires max 600 min 60



application
  service CME_AA flash://its-CISCO.2.0.1.0.tcl
  param operator 1999
  paramspace english language en
  paramspace english index 0
  paramspace english location flash://
  paramspace english prefix en
  param aa-pilot 5999



class-map match-all VoiceOverIPSignaling
 match ip dscp af31
class-map match-all VoiceOverIP
 match ip dscp ef
 match protocol sip
 match protocol skinny

policy-map VoiceOverIPPolicy
 class VoiceOverIP
  priority percent 10
 class VoiceOverIPSignaling
  bandwidth percent 2
 class class-default
  fair-queue


interface Dialer1
service-policy output VoiceOverIPPolicy

tftp-server flash:P00307020200.bin alias P00307020200.bin
tftp-server flash:P00307020200.loads alias P00307020200.loads
tftp-server flash:P00307020200.sb2 alias P00307020200.sb2
tftp-server flash:P00307020200.sbn alias P00307020200.sbn
tftp-server flash:Analog1.raw alias Analog1.raw
tftp-server flash:Analog2.raw alias Analog2.raw
tftp-server flash:AreYouThere.raw alias AreYouThere.raw
tftp-server flash:AreYouThereF.raw alias AreYouThereF.raw
tftp-server flash:Bass.raw alias Bass.raw
tftp-server flash:CallBack.raw alias CallBack.raw
tftp-server flash:Chime.raw alias Chime.raw
tftp-server flash:Classic1.raw alias Classic1.raw
tftp-server flash:Classic2.raw alias Classic2.raw
tftp-server flash:ClockShop.raw alias ClockShop.raw
tftp-server flash:DistinctiveRingList.xml alias DistinctiveRingList.xml
tftp-server flash:Drums1.raw alias Drums1.raw
tftp-server flash:Drums2.raw alias Drums2.raw
tftp-server flash:FilmScore.raw alias FilmScore.raw
tftp-server flash:HarpSynth.raw alias HarpSynth.raw
tftp-server flash:Jamaica.raw alias Jamaica.raw
tftp-server flash:KotoEffect.raw alias KotoEffect.raw
tftp-server flash:MusicBox.raw alias MusicBox.raw
tftp-server flash:Piano1.raw alias Piano1.raw
tftp-server flash:Piano2.raw alias Piano2.raw
tftp-server flash:Pop.raw alias Pop.raw
tftp-server flash:Pulse1.raw alias Pulse1.raw
tftp-server flash:Ring1.raw alias Ring1.raw
tftp-server flash:Ring2.raw alias Ring2.raw
tftp-server flash:Ring3.raw alias Ring3.raw
tftp-server flash:Ring4.raw alias Ring4.raw
tftp-server flash:Ring5.raw alias Ring5.raw
tftp-server flash:Ring6.raw alias Ring6.raw
tftp-server flash:Ring7.raw alias Ring7.raw
tftp-server flash:RingList.xml alias RingList.xml
tftp-server flash:Sax1.raw alias Sax1.raw
tftp-server flash:Sax2.raw alias Sax2.raw
tftp-server flash:Vibe.raw alias Vibe.raw
tftp-server flash:NeoWestern.wav alias NeoWestern.wav



voice-port 1/1/0
 supervisory disconnect dualtone mid-call
 pre-dial-delay 0
 cptone JP
 timeouts call-disconnect 0
 timeouts ringing 45
 timeouts wait-release 2
 connection plar 5999



ccm-manager music-on-hold


dial-peer voice 1 voip
 destination-pattern 1...
 session target ipv4:10.0.0.1

dial-peer voice 2 voip
 destination-pattern 2...
 session target ipv4:10.0.208.1

dial-peer voice 99 pots
 destination-pattern .T
 port 1/1/0
 forward-digits all

dial-peer voice 5999 voip
 service cme_aa
 destination-pattern 5999
 session target ipv4:172.16.0.1
 incoming called-number 5999
 dtmf-relay h245-alphanumeric
 codec g711ulaw
 no vad

sip-ua



telephony-service
 load 7960-7940 P00307020200
 max-ephones 24
 max-dn 48
 ip source-address 10.0.0.1 port 2000
 service phone displayIdleTimeout 00:30
 service phone displayOnDuration 1:00
 timeouts interdigit 2
 system message *****
 url services http://phone-xml.berbee.com/menu.xml
 time-zone 44
 time-format 24
 create cnf-files version-stamp 7960 Aug 28 2010 23:43:17
 max-conferences 4 gain -6
 call-forward pattern ....
 moh music-on-hold.au
 web admin system name admin secret *****
 dn-webedit
 transfer-system full-consult
 transfer-pattern ....
 secondary-dialtone 99
 after-hours block pattern 1 1900....... 7-24
 after-hours block pattern 2 0990...... 7-24
 directory entry 2 ***** name *****
 directory entry 1 ***** name *****




ephone-template  1
 softkeys idle  Redial Newcall Pickup Cfwdall Dnd
 softkeys seized  Redial Endcall Cfwdall Pickup Gpickup
 softkeys alerting  Endcall Callback
 softkeys connected  Hold Confrn Flash Park Trnsfer



ephone-dn  1  dual-line
 call-waiting ring
 number 1009
 pickup-group 1
 label *****
 description *****
 name *****
 call-forward busy 1599
 call-forward noan 1599 timeout 45


ephone-dn  2  dual-line
 call-waiting ring
 number 1001
 pickup-group 1
 label *****
 description *****
 name *****
 call-forward busy 1599
 call-forward noan 1599 timeout 45


ephone-dn  3  dual-line
 call-waiting ring
 number 2001
 pickup-group 2
 label *****
 description *****
 name *****
 call-forward busy 1599
 call-forward noan 1599 timeout 45


ephone-dn  4  dual-line
 call-waiting ring
 number 1002
 pickup-group 1
 label *****
 description *****
 name *****
 call-forward busy 1599
 call-forward noan 1599 timeout 45


ephone-dn  5  dual-line
 call-waiting ring
 number 2002
 pickup-group 2
 label *****
 description *****
 name *****
 call-forward busy 1599
 call-forward noan 1599 timeout 45





ephone-dn  9
 number 9999
 paging ip 239.1.1.100 port 2000


ephone-dn  10
 number 1999


ephone-dn  11
 number 2999


ephone-dn  40
 number 5001
 park-slot timeout 90 limit 3


ephone-dn  41
 number 5002
 park-slot timeout 90 limit 3


ephone-dn  42
 number 5003
 park-slot timeout 90 limit 3

ephone-dn  47
 number A2
 intercom A1 barge-in no-mute label "*****"


ephone-dn  48
 number A1
 intercom A2 barge-in no-mute label "*****"


ephone  1
 description *****
 ephone-template 1
 mac-address *****
 paging-dn 9
 keep-conference
 button  1o1,10 6:47

ephone  2
 description *****
 ephone-template 1
 mac-address *****
 paging-dn 9
 keep-conference
 button  1o2,10 6:48

ephone  3
 description *****
 ephone-template 1
 mac-address *****
 paging-dn 9
 keep-conference
 button  1o3,11

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Question by:usslindstrom
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9 Comments
 
LVL 5

Author Comment

by:usslindstrom
ID: 33595330
*BTW - I meant to say "limit the number of inbound phone RINGS"...   I mispoke and said "calls".   I don`t want to limit those,  calls are good.  ;)
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LVL 9

Assisted Solution

by:Alex Bahar
Alex Bahar earned 2000 total points
ID: 33603648
timeouts call-disconnect
Specifies the timeout value for releasing an FXO voice port when an incoming call is not answered.  
Your configured value of 0 tells the router  to ring the phone forever as it disables disconnect supervision. Try changing it to a normal value like 60.
0
 
LVL 9

Assisted Solution

by:Alex Bahar
Alex Bahar earned 2000 total points
ID: 33603682
Your park-slot config is correct (you can't make mistakes there anyway).
When you park a call, the park number should be displayed on the phone. And you dial that number from another phone to pick up the parked call. Does the parked number display correctly on the reception phone?
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LVL 5

Author Comment

by:usslindstrom
ID: 33605475
Thank you for the suggestions.  But unfortunately - they're "no go"...

My automated attendent phone line is 5999.  When a call comes in over POTS - and then directed via the automated setup, the IP phones show that the call is coming from that DN.

If the IP phone does not pickup before the POTS phone disconnects, it will stop ringing after a few seconds...  wait for about 10-15 then ring again continuously until the phone is picked up.  On the phone display, it shows a call from the automated 5999 number.

Of course when the user finally does pick up the IP phone, the call immediately disconnects, since the POTS user is no longer calling.

--------------------

On the call parking problem.  I will test pickup via another phone when I have a second, but from what I"m noticing, is the user that parked the call - is the one receiving the busy signal when  they try to dial the parked number.

Is this by design then?    Just wondering.





Thanks for all the assistance so far.   Playing with phones makes me wanna' drop my routing and switching path and get through the voice certs.  :)
0
 
LVL 5

Author Comment

by:usslindstrom
ID: 33605477
Oh, and I did implement your command line w/45 sec delay:

timeouts call-disconnect 45
0
 
LVL 9

Assisted Solution

by:Alex Bahar
Alex Bahar earned 2000 total points
ID: 33605768
>My automated attendent phone line is 5999.  When a call comes in over POTS - and then directed via the automated setup, the IP phones show that the call is coming from that DN.

It seems the callerid does not work well on your FXO. Have you asked your carrier to deliver Caller ID on this line? You can verify it by connecting an ordinary phone to one of your external lines.
Callerid can be delivered by using a few different methods like FSK, DTMF etc. It also depends on the country you live in. Fox example in Australia only an FXO M3 type card can collect the callerid. This is an old technology pushed to its limits, and getting it working and doing advanced stuff required a lot of try and learn.
http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
 
>If the IP phone does not pickup before the POTS phone disconnects, it will stop ringing after a few seconds...  wait for about 10-15 then ring again continuously until the phone is picked up.
Hmm this is a strange behaviour. I am suspecting either a conflicting supervision message (for answer, disconnect) or some other issue that is causing your auto attendant script to regenerate the call when caller hangs up. Have you tried to turn on debugs for autoattendant calls and try to see what triggers the second ring?
Troubleshooting FXO is a real challenge. Because of that reason most people go with ISDN BRI which allows you to use DID, and troubleshooting is much easier because of its standardised digital Q931 protocol. Unless this is a very expensive option, I would suggest you to consider switching over to ISDN BRI.
0
 
LVL 5

Author Comment

by:usslindstrom
ID: 33606015
You most likely hit it on the head with your comments there.

I'm in Japan, using NTT's fiber plan.  - And have spent many many many troubleshooting nights on their ADSL v2 crap that couldn't train with ANY of my cisco equipment (HWIC, WIC-1ADSL, etc).  I noticed on your link that Japan isn't mentioned either.  I do have the VIC-2FXO card in a NM-2V on my 2620xm.

The caller ID issue that you mentioned is almost guaranteed that they use some proprietary protocol on their SIP implementation.  (The FXO port is plugged into a VOIP box supplied by them).  - And that's how I get my external calls.  Everything works great, with the exception of what I've been mentioning above.

Grabbing another ISP (ISDN BRI) option currently is out of the way - (already spending past the budgeted amount to keep things operational this year).

Is there a way to implement that command "timeouts call-disconnect 45" on voip-to-voip calls? - Instead of on the interface of the FXO port?

Again, thanks for giving me all this information.  It's much appreciated.
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LVL 9

Accepted Solution

by:
Alex Bahar earned 2000 total points
ID: 33606237
> Is there a way to implement that command "timeouts call-disconnect 45" on voip-to-voip calls? - Instead of on the interface of the FXO port?
Unfortunately this command only applies to analog ports.
BTW have you tried to set call-disconnect timeout to 1 second? ( Unfortunately sometimes it is try and learn with the FXO)
A few more points>
Make sure you configured "cptone jp" on your analog ports.
Verify the phone wire polarity is correct. If debug vpm signal command outputs NO_TIP_GROUND that means the polarity of the RJ-11 cables is mismatched. Human speech is not affected by polarity mismatches, but it can cause disconnect supervision issues.
0
 
LVL 5

Author Closing Comment

by:usslindstrom
ID: 33608213
Thank you very much for the information.  It looks like it's just gonna' have to be plug and play with these settings.  I appreciate all the help you've shown me.

I'll keep toying away with all the configs and see if something can't come of any changes.
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