Call Manager Express - Small Configuration Issues
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.
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
SOLUTION
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ASKER
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. :)
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. :)
ASKER
Oh, and I did implement your command line w/45 sec delay:
timeouts call-disconnect 45
timeouts call-disconnect 45
SOLUTION
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ASKER
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.
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.
ASKER CERTIFIED SOLUTION
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ASKER
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.
I'll keep toying away with all the configs and see if something can't come of any changes.
ASKER