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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
  • 5
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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.  ;)
0
 
LVL 9

Assisted Solution

by:Alex Bahar
Alex Bahar earned 500 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.
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Assisted Solution

by:Alex Bahar
Alex Bahar earned 500 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.  :)
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LVL 5

Author Comment

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

timeouts call-disconnect 45
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Assisted Solution

by:Alex Bahar
Alex Bahar earned 500 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.
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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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Accepted Solution

by:
Alex Bahar earned 500 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.
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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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