FXO port issue

Dear all,
Am using cisco 2921 CME 8.6 with ios version c2900-universalk9-mz.SPA.151-4.M4.bin..
Also am using a dedicated one fxo port for a specific user to send and receive calls,, the user always faces busy tone when initiate outgoing call, the fxo port doesnt disconnect from the previous call.
voice-port 0/3/2
supervisory disconnect dualtone mid-call
no battery-reversal
cptone SA
timeouts interdigit 3
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 4015
impedance complex2
caller-id enable
Any Idea to solve the issue, please???
#sh voice port 0/3/2
Foreign Exchange Office 0/3/2 Slot is 0, Sub-unit is 3, Port is 2
Type of VoicePort is FXO
Operation State is DORMANT
Administrative State is UP
The Last Interface Down Failure Cause is Administrative Shutdown
Description is not set
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 128 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is plar opx
Connection Number is 4015
Initial Time Out is set to 15 s
Interdigit Time Out is set to 3 s
Call Disconnect Time Out is set to 1 s
Power Denial Disconnect Time Out is set to 1000 ms
Ringing Time Out is set to 180 s
Wait Release Time Out is set to 1 s
Companding Type is u-law
Region Tone is set for SA
Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to complex2 Ohm
Station name None, Station number None
Caller ID Info Follows:
Standard BELLCORE
Caller ID is received after 1 ring(s)
Translation profile (Incoming):
Translation profile (Outgoing):
lpcor (Incoming):
lpcor (Outgoing):
Voice card specific Info Follows:
Signal Type is loopStart
Battery-Reversal is disabled
Number Of Rings is set to 1
Supervisory Disconnect is dualtone mid-call
Answer Supervision is inactive
Hook Status is On Hook
Ring Detect Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Dial Out Type is dtmf
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Pulse Rate Timing is set to 10 pulses/second
InterDigit Pulse Duration Timing is set to 750 ms
Percent Break of Pulse is 65 percent
GuardOut timer is 2000 ms
Minimum ring duration timer is 125 ms
Hookflash-in Timing is set to 600 ms
Hookflash-out Timing is set to 400 ms
Supervisory Disconnect Timing (loopStart only) is set to 350 ms
OPX Ring Wait Timing is set to 6000 ms
Secondary dialtone is disabled

Hi again Lan,
Strange issue occur, the voice ports seem to be ok, in on-hook status and incoming calls are fine, the strange issue is outgoing calls can`t be initiated despite the dial-peer are configured. this issue happens to me suddenly before changing any thing. As soon i dial the first digit after dialing 9, i receive a fast busy tone, like below as soon i dial 7 i receive a fast busy tone from an ip phone with no restriction...
dial-peer voice 2002 pots
trunkgroup 2
corlist outgoing CDMA
destination-pattern 977.......
forward-digits 9
voice-port 0/2/0
trunk-group 2
supervisory disconnect dualtone mid-call
no battery-reversal
cptone SA
timeouts call-disconnect 1
timeouts wait-release 1
connection plar opx immediate 4446
impedance complex2
caller-id enable
I restart the router, boot from another ios, remove and configure the dial-peers, shut and unshut the voice port, nothing solve the issue. it seems it doesn`t recognize outside digits,,, local calls are working fine..
Any ideas to solve the issue please...
the below is info you requested while the ports are hanging:
VoiceGW#sh voice call status
CallID     CID ccVdb      Port        Slot/DSP:Ch Called #   Codec    MLPP Dial-peers
0x11       121D 0x2B326244 0/2/1            No DSP 7444624261 None     0/1015
0x17       121D 0x2B334ADC 0/3/3            No DSP *7444624261 None      1015/0
1 active call found
VoiceGW#sh voice call status
Enter configuration commands, one per line. End with CNTL/Z.
VoiceGW(config)#exdo sh voice port summ
                                           IN       OUT
PORT            CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
=============== == ============ ===== ==== ======== ======== ==
0/1/0           -- fxs-ls      up    dorm on-hook idle     y
0/1/1           -- fxs-ls      up    dorm on-hook idle     y
0/1/2           -- fxs-ls      up    dorm on-hook idle     y
0/1/3           -- fxs-ls      up    dorm on-hook idle     y
0/2/0           -- fxo-ls      up    dorm idle     on-hook y
0/2/1           -- fxo-ls      up    up   idle     off-hook y
0/2/2           -- fxo-ls      up    dorm idle     on-hook y
0/2/3           -- fxo-ls      up    dorm idle     on-hook y
0/3/0           -- fxo-ls      up    dorm idle     on-hook y
0/3/1           -- fxo-ls      up    dorm idle     on-hook y
0/3/2           -- fxo-ls      up    dorm idle     on-hook y
0/3/3           -- fxo-ls      up    up   idle     off-hook y
50/0/1          1      efxs     up    up   on-hook idle     y
50/0/1          2      efxs     up    up   on-hook idle     y
VoiceGW(config-voiceport)#no sh
Jul 31 12:13:57.923: htsp_process_event: [0/2/1, FXOLS_CONNECT, E_HTSP_IF_OOS]htspm_mgt_statehtspm_disc_ind
Jul 31 12:13:57.923: htsp_timer_stop
Jul 31 12:13:57.923: htsp_timer_stop2
Jul 31 12:13:57.923: htsp_timer - 1000 msec
Jul 31 12:13:57.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_RELEASE_REQ]act_disc_conf
Jul 31 12:13:57.923: htsp_process_event: [0/3/3, FXOLS_OFFHOOK, E_HTSP_RELEASE_REQ]fxols_offhook_release
Jul 31 12:13:57.923: htsp_timer_stop
Jul 31 12:13:57.923: htsp_timer_stop2
QNB_VoiceGW(config-voiceport)#Jul 31 12:13:57.923: htsp_timer_stop3
Jul 31 12:13:57.923: [0/3/3] set signal state = 0x4 timestamp = 0
Jul 31 12:13:57.923: htsp_timer - 2000 msec
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_EVENT_TIMER]htspm_disc_conf
Jul 31 12:13:58.923: htsp_timer_stop
Jul 31 12:13:58.923: htsp_timer_stop2
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_IF_OOS_CONF]
Jul 31 12:13:58.923: htsp_timer_stop
Jul 31 12:13:58.923: htsp_timer_stop2
Jul 31 12:13:58.923: htsp_timer_stop3
Jul 31 12:13:58.923: htsp_timer_stop_mlpp
Jul 31 12:13:58.923: %LINK-3-UPDOWN: Interface Foreign Exchange Office 0/2/1, changed state to Administrative Shutdown
Jul 31 12:13:58.923: htsp_process_event: [0/2/1, S_DOWN, E_DSP_INTERFACE_INFO]
Jul 31 12:13:59.339: htsp_process_event: [0/2/1, S_DOWN, E_HTSP_IF_INSERVICE]
Jul 31 12:13:59.343: %LINK-3-UPDOWN: Interface Foreign Exchange Office 0/2/1, changed state to up
Jul 31 12:13:59.343: Foreign Exchange Office 0/2/1 rx_signal_map:
F F F F
5 F F F
F F F F
F F F F
Jul 31 12:13:59.343: [0/2/1] set signal state = 0xC timestamp = 0
Jul 31 12:13:59.343: Foreign Exchange Office 0/2/1 tx_signal_map:
0 4 4 4
4 4 6 4
C C C C
C C C C
Jul 31 12:13:59.343: htsp_process_event: [0/2/1, S_OPEN_PEND, E_HTSP_GO_UP]
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_NULL, E_HTSP_INIT]fxols_null_init
Jul 31 12:13:59.455: [0/2/1] set signal state = 0xC timestamp = 0
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_INIT, E_HTSP_INSERVE]fxols_init_inserve
Jul 31 12:13:59.455: [0/2/1] set signal state = 0x4 timestamp = 0
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jul 31 12:13:59.455: htsp_process_event: [0/2/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jul 31 12:13:59.923: htsp_process_event: [0/3/3, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Jul 31 12:13:59.927: htsp_process_event: [0/3/3, FXOLS_ONHOOK, E_DSP_SIG_0100]

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! Last configuration change at 13:36:42 ZP4 Thu Sep 13 2012 by Nick
! NVRAM config last updated at 13:45:41 ZP4 Thu Sep 13 2012 by Nick
version 15.1
parser config cache interface
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
service internal
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hostname UC540
boot-start-marker
boot system flash:uc500-advipservicesk9-mz.151-2.T4
boot-end-marker
logging buffered 64000
enable secret 5 $1$3CIf$.rXyHeJQrwd97X/f2dS0M1
no aaa new-model
clock timezone ZP4 4 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-3558175224
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-3558175224
revocation-check none
crypto pki certificate chain TP-self-signed-3558175224
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dot11 syslog
dot11 ssid cisco-data
vlan 1
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ip source-route
ip cef
ip dhcp relay information trust-all
ip dhcp excluded-address 10.1.3.1 10.1.3.10
ip dhcp pool phone
   network 10.1.3.0 255.255.255.0
   default-router 10.1.3.1
   option 150 ip 10.1.3.1
ip name-server 213.42.20.20
ip name-server 195.229.241.222
ip inspect WAAS flush-timeout 10
ip inspect name SDM_LOW cuseeme
ip inspect name SDM_LOW dns
ip inspect name SDM_LOW ftp
ip inspect name SDM_LOW h323
ip inspect name SDM_LOW https
ip inspect name SDM_LOW icmp
ip inspect name SDM_LOW imap
ip inspect name SDM_LOW pop3
ip inspect name SDM_LOW netshow
ip inspect name SDM_LOW rcmd
ip inspect name SDM_LOW realaudio
ip inspect name SDM_LOW rtsp
ip inspect name SDM_LOW esmtp
ip inspect name SDM_LOW sqlnet
ip inspect name SDM_LOW streamworks
ip inspect name SDM_LOW tftp
ip inspect name SDM_LOW tcp router-traffic
ip inspect name SDM_LOW udp router-traffic
ip inspect name SDM_LOW vdolive
no ipv6 cef
multilink bundle-name authenticated
stcapp ccm-group 1
stcapp
stcapp supplementary-services
port 0/0/0
fallback-dn 301
port 0/0/1
fallback-dn 302
port 0/0/2
fallback-dn 303
port 0/0/3
fallback-dn 304
trunk group ALL_FXO
max-retry 5
voice-class cause-code 1
hunt-scheme longest-idle
translation-profile outgoing PROFILE_ALL_FXO
trunk group ALL_FX0
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voice rtp send-recv
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
sip
no update-callerid
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g711ulaw
voice class dualtone-detect-params 1
freq-max-deviation 50
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 6
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 406
cadence 398 344 237 527 400
voice class custom-cptone CCAjointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone CCAleavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
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voice register global
voice hunt-group 1 parallel
list 301,302,303
timeout 24
pilot 511
voice translation-rule 4
rule 15 // //
voice translation-rule 1000
rule 1 /.*/ //
voice translation-rule 1111
voice translation-rule 1112
rule 1 /^9/ //
rule 3 /^0/ //
voice translation-rule 2222
voice translation-rule 3265
rule 1 /$^..........$$/ /9\1/
rule 2 /$^.........$$/ /9\1/
rule 15 /$^ABCD$$/ /ABCD\1/
voice translation-profile CALLER_ID_TRANSLATION_PROFILE
translate calling 1111
voice translation-profile CallBlocking
translate called 2222
voice translation-profile INCOMING_CallerID_PROFILE
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voice translation-profile OUTGOING_TRANSLATION_PROFILE
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voice translation-profile PROFILE_ALL_FXO
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voice translation-profile nondialable
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dspfarm
dsp services dspfarm
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archive
log config
logging enable
logging size 600
hidekeys
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username adminac privilege 15 secret 5 $1$NDC.$PtD0y4YGIj5SqI1gghxWE1
username Nick privilege 15 secret 5 $1$iAmL$tsg7Jf2TEND1NN.h8z2dy/
ip tftp source-interface Loopback0
bridge irb
interface Loopback0
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ip address 10.1.10.2 255.255.255.252
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ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
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ip address 192.168.101.2 255.255.255.252
ip nat outside
ip virtual-reassembly in
duplex auto
speed auto
interface Integrated-Service-Engine0/0
description cue is initialized with default IMAP group
ip unnumbered Loopback0
ip nat inside
ip virtual-reassembly in
service-module ip address 10.1.10.1 255.255.255.252
service-module ip default-gateway 10.1.10.2
interface FastEthernet0/1/0
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/1
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/2
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/3
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/4
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/5
switchport voice vlan 100
macro description cisco-phone
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macro description cisco-phone
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switchport access vlan 20
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bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
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no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
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access-list 1 permit 192.168.10.0 0.0.0.255
access-list 1 permit 10.1.3.0 0.0.0.255
access-list 1 permit 10.1.10.0 0.0.0.3
access-list 100 remark auto generated by SDM firewall configuration
access-list 100 remark SDM_ACL Category=1
access-list 100 deny   ip 192.168.10.0 0.0.0.255 any
access-list 100 deny   ip host 255.255.255.255 any
access-list 100 deny   ip 127.0.0.0 0.255.255.255 any
access-list 100 permit ip any any
access-list 101 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 101 remark SDM_ACL Category=1
access-list 101 permit tcp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 permit udp 10.1.3.0 0.0.0.255 eq 2000 any
access-list 101 deny   ip 10.1.3.0 0.0.0.255 any
access-list 101 deny   ip 192.168.10.0 0.0.0.255 any
access-list 101 deny   ip 192.168.101.0 0.0.0.3 any
access-list 101 deny   ip host 255.255.255.255 any
access-list 101 deny   ip 127.0.0.0 0.255.255.255 any
access-list 101 permit ip any any
access-list 102 remark auto generated by SDM firewall configuration##NO_ACES_6##
access-list 102 remark SDM_ACL Category=1
access-list 102 deny   ip 10.1.10.0 0.0.0.3 any
access-list 102 deny   ip 10.1.3.0 0.0.0.255 any
access-list 102 deny   ip 192.168.101.0 0.0.0.3 any
access-list 102 deny   ip host 255.255.255.255 any
access-list 102 deny   ip 127.0.0.0 0.255.255.255 any
access-list 102 permit ip any any
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access-list 103 deny   ip 192.168.101.0 0.0.0.3 any
access-list 103 deny   ip host 255.255.255.255 any
access-list 103 deny   ip 127.0.0.0 0.255.255.255 any
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bridge 100 route ip
voice-port 0/0/0
cptone GB
station-id name Cordless
station-id number 329
caller-id enable
voice-port 0/0/1
cptone AE
caller-id enable
voice-port 0/0/2
cptone AE
caller-id enable
voice-port 0/0/3
cptone AE
caller-id enable
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trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4FXO-0/1/0-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/1
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/1-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/2
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
supervisory dualtone-detect-params 1
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/2-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/1/3
trunk-group ALL_FX0 64
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone GB
connection plar opx 511
impedance 600c
description Configured by CCA 4 FXO-0/1/3-Custom-BG
bearer-cap Speech
caller-id enable
voice-port 0/4/0
auto-cut-through
signal immediate
input gain auto-control -15
description Music On Hold Port
sccp local Loopback0
sccp ccm 10.1.3.1 identifier 1 version 4.0
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register confprof1
dspfarm profile 1 conference
description DO NOT MODIFY, active CCA conference profile - CCA2.0 codec729
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
dial-peer cor custom
name internal
name local
name local-plus
name international
name national
name national-plus
name emergency
name toll-free
dial-peer cor list call-internal
member internal
dial-peer cor list call-local
member local
dial-peer cor list call-local-plus
member local-plus
dial-peer cor list call-national
member national
dial-peer cor list call-national-plus
member national-plus
dial-peer cor list call-international
member international
dial-peer cor list call-emergency
member emergency
dial-peer cor list call-toll-free
member toll-free
dial-peer cor list user-internal
member internal
member emergency
dial-peer cor list user-local
member internal
member local
member emergency
member toll-free
dial-peer cor list user-local-plus
member internal
member local
member local-plus
member emergency
member toll-free
dial-peer cor list user-national
member internal
member local
member local-plus
member national
member emergency
member toll-free
dial-peer cor list user-national-plus
member internal
member local
member local-plus
member national
member national-plus
member emergency
member toll-free
dial-peer cor list user-international
member internal
member local
member local-plus
member international
member national
member national-plus
member emergency
member toll-free
dial-peer voice 1 pots
port 0/0/0
no sip-register
dial-peer voice 2 pots
port 0/0/1
no sip-register
dial-peer voice 3 pots
port 0/0/2
no sip-register
dial-peer voice 4 pots
port 0/0/3
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dial-peer voice 5 pots
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destination-pattern ABC
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no sip-register
dial-peer voice 50 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/0
dial-peer voice 51 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/1
dial-peer voice 52 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/2
dial-peer voice 53 pots
description ** incoming dial peer **
incoming called-number ^AAAA$
port 0/1/3
dial-peer voice 54 pots
description ** FXO pots dial-peer **
destination-pattern A0
port 0/1/0
no sip-register
dial-peer voice 55 pots
description ** FXO pots dial-peer **
destination-pattern A1
port 0/1/1
no sip-register
dial-peer voice 56 pots
description ** FXO pots dial-peer **
destination-pattern A2
port 0/1/2
no sip-register
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
destination-pattern 388
b2bua
session protocol sipv2
session target ipv4:10.1.10.1
voice-class sip outbound-proxy ipv4:10.1.10.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
dial-peer voice 6 pots
description "catch all dial peer for BRI/PRI"
translation-profile incoming nondialable
incoming called-number .%
direct-inward-dial
dial-peer voice 57 pots
description ** FXO pots dial-peer **
destination-pattern A3
port 0/1/3
no sip-register
dial-peer voice 69 pots
destination-pattern 329
port 0/0/0
dial-peer voice 300 pots
trunkgroup ALL_FX0
description Local Numbers
destination-pattern 9T
forward-digits 9
dial-peer voice 301 voip
destination-pattern 2..
session target ipv4:192.168.201.2
dial-peer voice 303 pots
trunkgroup ALL_FXO
trunkgroup ALL_FX0
description **InternationalCall**
destination-pattern 88T
dial-peer voice 304 pots
trunkgroup ALL_FX0
description *EM1*
destination-pattern 9[1-9]T
forward-digits 3
dial-peer voice 302 pots
trunkgroup ALL_FX0
description **Mobiles**
destination-pattern 9.[0-9].[0-9]......
dial-peer voice 305 pots
trunkgroup ALL_FX0
description **800-**
destination-pattern 9[0-9][0-9][0-9]T
no dial-peer outbound status-check pots
telephony-service
sdspfarm conference mute-on 111 mute-off 222
sdspfarm units 5
sdspfarm tag 1 confprof1
conference hardware
video
fxo hook-flash
max-ephones 40
max-dn 300
ip source-address 10.1.3.1 port 2000
max-redirect 20
auto assign 1 to 1 type bri
calling-number initiator
service phone videoCapability 1
service phone webAccess 0
service dnis overlay
service dnis dir-lookup
timeouts interdigit 5
system message American Center
url services http://10.1.10.1/voiceview/common/login.do
url authentication http://10.1.10.2/CCMCIP/authenticate.asp
load 521G-524G cp524g-8-1-17
load 525G spa525g-7-4-8
load 501G spa5x5-7-1-3c
load 502G spa5x5-7-1-3c
load 504G spa5x5-7-1-3c
load 508G spa5x5-7-1-3c
load 509G spa5x5-7-1-3c
time-zone 35
date-format dd-mm-yy
voicemail 388
max-conferences 8 gain -6
call-forward pattern .T
call-forward system redirecting-expanded
hunt-group logout HLog
moh MOH2.wav
multicast moh 239.10.16.16 port 2000
web admin system name cisco secret 5 $1$iDgA$MKNi2RWfsO0KjuC82kgLJ1
dn-webedit
time-webedit
transfer-system full-consult dss
transfer-pattern 9.T
transfer-pattern .T
secondary-dialtone 9
fac standard
create cnf-files version-stamp 7960 Aug 29 2012 12:00:04
line con 0
privilege level 15
logging synchronous
no modem enable
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
line vty 0 4
exec-timeout 0 0
logging synchronous
login local
transport input all
line vty 5 100
login local
transport input all
ntp master
end
Some of the output are not shown becaus it is to long I have attach the whole config for reference and any advice on how could I optimize and resolve my issues is greatly appreciated. Thanks

Nicolo - First off this stuff gets crazy sometimes. No worries about the exam. Sometimes when FXO ports go crazy it is due to battery reversal. If you go to the FXO port settings try turning battery reversal on and or off... depending on its current setting. See if that helps.
As for the 525s not registering.. These are inside the network correct? Are you connecting one directly to the UC500 with a Cat5E or Cat6 patch cable and the same thing happens? Does the MAC address on the phone match a MAC address under the EPHONE settings?
If you telnet into the UC500 can you execute a "dir" command at the CLI prompt and "CD" (change directory) into the phones folder and then the spa525g folder? Do files exist in there?
Also I only see an IP address under BVI100? This is the voice side of things what happened to the IP address under BVI1 (Data VLAN). Can you give us some information about the internal network? Cna you PING this phone system from the network? What IP address does it have?

Issue connecting SPA3102 FXO port to paging interface

Hello,
I was referred to this community after a call to small business support, we were unable to come to a solution.
SPA3102 v5.2.13 FW
Viking Electronics PI-1 Paging Interface
We were told from Viking that the interface only supports an FXO port, all the connections were verified with their support and we're able to page by connecting an analog phone straight to the interface. Plugging into the FXS port on the SPA3102 also results in a dial tone playing over the paging system.
All settings on the SPA3102 are default minus SIP registration information, we see the unit as registered within the web interface as well as within our Asterisk CLI.
Trying to ring the extension results in constant ringing if voicemail is disabled, and straight to voicemail if enabled. Within the Asterisk CLI we see a SIP response 503 "Service Unavailable."
I can get the SPA3102 to respond by lowering the "Line In Use Voltage" down from its default value of 30. However all I hear is a hum and I'm not able to speak over the paging system.
Any insight into this issue would be greatly appreciated, we believe it's just settings on the SPA3102 unit itself that need to be changed.
Thanks!

Looking at the Viking Pl-1 web page installation instructions, I think the key setting on the SPA3102 would be to set the voip-to-pstn gateway dial plan to NONE. The "hum" you get when you lower the Line-In-Use setting sounds like the dial tone you would get with the default dial plan setting.
I would try the following:
Viking Pl-1:
Talk Battery Switch (PT): On
Audio In: Cabled to FXO port of SPA3102
SPA3102:
Interface Analog Phone attached to SPA3102:
Line 1 Tab
Line Enable: YES
Dial Plan: (S0<:@gw0>)
PSTN Line Tab
Line Enable: Yes
VoIP-to-PSTN Gateway Enable: Yes
One Stage Dialing: Yes
Line 1 VoIP Caller DP: NONE
VoIP Answer Delay: 0
Line In Use Voltage: xx
The Line In Use Voltage needs to be lower than the Talk Battery voltage supplied by the Viking Pl-1. Usually it is set about half way between the on-hook and off-hook voltage level. Measure the FXO (PSTN Line) on-hook voltage by reading it on the SPA3102 INFO Tab. Set the Line In Use Voltage substantially below the on-hook talk voltage.
If you lift the phone with the above settings you should be connected to the paging system.
Interface to an asterisk pbx system:
PSTN Line Tab
Setup Registration on PSTN Line Tab to Asterisk system.
Register: Yes
Proxy: xxx
UserID: xxx
Password: xxx
VoIP-to-PSTN Gateway Enable: Yes
VoIP Caller Auth Method: None
One Stage Dialing: Yes
VoIP Caller Default DP: NONE
VoIP Answer Delay: 0
Line In Use Voltage: same comments as above
If you call the extension on the asterisk PBX you should be attached to the Viking unit.

Incorrect Caller ID on calls from outside line via FXO port.

   Have a public phone line connected to my CUCME 2801 router VIC2-2FXO card. All inbound calls are passed to DN-5001 (group number). Can receive and send calls without a problem, but incoming calls all show "911" for caller ID. Think this is simply an issue with the out bound dial-peer, of which the lowest numbered out bound dial-peer is for 911 services. Not sure how to correct this so inbound calls show the proper caller ID?
    Below is a copy of my CUCME show run output from the FXO port config thru all the dial-peers. Any pointers is greatly appreciated.
    Thanks.
           Kirk E.
voice-port 0/0/0
connection plar opx immediate 5001
voice-port 0/0/1
voice-port 0/2/0
station-id name POTS
station-id number 7000
voice-port 0/2/1
ccm-manager config
dial-peer voice 7000 pots
destination-pattern 5006
port 0/2/0
dial-peer voice 90 pots
description Emergency Services
destination-pattern 911
port 0/0/0
forward-digits 3
dial-peer voice 91 pots
description 10 Digit local dialing
destination-pattern [234].........
port 0/0/0
forward-digits 10
dial-peer voice 92 pots
description 11 Digit local/long distance dialing
destination-pattern 1[2348].........
port 0/0/0
forward-digits 11
dial-peer voice 93 pots
description Long Distance
destination-pattern 011T
port 0/0/0
prefix 011
dial-peer voice 94 pots
description Backup bench POTS phone
destination-pattern 7000
port 0/2/0
dial-peer voice 2 voip
destination-pattern 51..
session protocol sipv2
session target ipv4:172.16.2.155
dtmf-relay sip-notify
codec g711ulaw
no vad

Hi
Can you find the below:-
Hi
1- Please find the below table as the following link http://www.cisco.com/en/US/products/hw/routers/ps274/products_tech_note09186a00800b53c7.shtml
Caller ID          Requires VIC-2FXO-M1, VIC-2FXO-M2, VIC-4FXO-M1, VIC2-2FXO, VIC2-4FXO, or MRP3-8FXOM1
under voice-port
caller-id enable
2-If above configure and still have no caller id , please add the below commannds to the voice-port
caller-id alerting line-reversal
cptone ?               "based on your"
caller-id alerting ring 2    "the default is 1" maximum number of rings to be detected before a call is answered over an FXO voice port.
4-Do debug to make sure all ok
"debug vpm signal "
[0/3/0] get_fxo_caller_id:Caller ID received. Message type=128 length=31 checksum=74
Thank you
please rate all useful information

CME fxo port problem

hi,
I have a 2801 CME and 2 fxo lines. Some times when both lines are seized (because of incoming and outgoing call) and after the phones are on hook the lines are still seized and are not released and when i removed the cables from fxo port, the led on each port start to blink in sequence and then when i reconnected, the port was busy again.. so i had to reboot the router to make it work. i changed the slot but dont think that it will cause any difference..do i need to upgrade the ios or some other issue..when i am calling the disconnect is fine but it happens some time when the line get stuck i have to reboot the router. one more thing when i changed the slot of the card the router didnt detected the change of slot and was showing fxo port on old slot. i rebooted the router to detect the fxo card on the right slot..sorry for the double post but no body was replying to i post it again..
Currently Being Moderated
May 12, 2012 11:49 AM (in response to Naresh Rathore)
FXO lines stuck/hang (2801 CME)
hi following is the sh run and sh version. cme output portion is removed
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2011.02.03 14:27:39 =~=~=~=~=~=~=~=~=~=~=~=
User Access Verification
Username: admin
Password:
Router_Home>en
Password:
Router_Home#term
Router_Home#terminal le
Router_Home#terminal length n   512
Router_Home#sh run
Building configuration...
Current configuration : 15203 bytes
! Last configuration change at 14:41:02 UTC Thu Feb 3 2011 by admin
! NVRAM config last updated at 14:38:09 UTC Thu Feb 3 2011 by admin
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router_Home
boot-start-marker
boot-end-marker
enable password cisco
no aaa new-model
clock calendar-valid
crypto pki trustpoint TP-self-signed-2416845307
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2416845307
revocation-check none
rsakeypair TP-self-signed-2416845307
crypto pki certificate chain TP-self-signed-2416845307
certificate self-signed 01
30820243 308201AC A0030201 02020101 300D0609 2A864886 F70D0101 04050030
31312F30 2D060355 04031326 494F532D 53656C66 2D536967 6E65642D 43657274
69666963 6174652D 32343136 38343533 3037301E 170D3131 30323032 31373332
34335A17 0D323030 31303130 30303030 305A3031 312F302D 06035504 03132649
4F532D53 656C662D 5369676E 65642D43 65727469 66696361 74652D32 34313638
34353330 3730819F 300D0609 2A864886 F70D0101 01050003 818D0030 81890281
8100DFAC E4312F46 0F2BF242 E55FE0C1 73DC95F2 B2844295 DA691CE3 D9A202B1
77B1B0C3 5AE3D936 C4D786DB 7CFA6624 024C6A82 A29B3AAC 3BA89A77 5425A97A
6D79A88A C1327171 C88AA5E0 AB52F461 87FB472E A7622955 17C8F22C 58842EF4
4DFA422A 54E6B96A FA536C59 BD93FDCD 872C0586 08117535 2D13F1E0 A53E65AB
FE470203 010001A3 6B306930 0F060355 1D130101 FF040530 030101FF 30160603
551D1104 0F300D82 0B526F75 7465725F 486F6D65 301F0603 551D2304 18301680
14E940B0 E437790B 4B825CD2 0FA9020F 63C9ED3A ED301D06 03551D0E 04160414
E940B0E4 37790B4B 825CD20F A9020F63 C9ED3AED 300D0609 2A864886 F70D0101
04050003 8181006B D6136D19 6EF4DDCD B3AF591E 57B9B831 79578799 03862FCF
4AF772DE AC72FC85 3F6B6B20 81F528F0 F7B2CBD0 E9795060 C46AB102 AE2CDF53
11C39D67 B49A7AE8 FB619A0F 525543F7 8BA1D52C CABEFFEB 9E5EC7E7 938AA602
84F1ECD2 303E8609 A9AB0699 9078051B 1853BC9A 4B45F2A2 204310D5 8B34B5DD
2FA51064 EFF5D4
   quit
dot11 syslog
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 10.168.0.1
ip dhcp excluded-address 10.168.0.2
ip dhcp excluded-address 10.168.10.1
ip dhcp excluded-address 10.168.10.2
ip dhcp excluded-address 10.168.111.1 10.168.111.6
ip dhcp pool Data
   network 10.168.0.0 255.255.255.0
   default-router 10.168.0.1
   option 150 ip 10.168.10.1
   dns-server 212.72.1.186 212.72.23.4
ip dhcp pool VOICE
   network 10.168.10.0 255.255.255.0
   default-router 10.168.10.1
   option 150 ip 10.168.10.1
ip dhcp pool WLAN
   network 10.168.111.0 255.255.255.0
   default-router 10.168.111.1
   option 150 ip 10.168.10.1
   dns-server 212.72.1.186 212.72.23.4
ip host members.dyndns.org 204.13.248.112
ip name-server 212.72.1.186
ip name-server 212.72.23.4
ip ddns update method example_dyndns
HTTP
add http://whatisthis:[email protected]/nic/update?system=dyndns&hostname=<h>&myip=<a>
interval maximum 28 0 0 0
interval minimum 28 0 0 0
multilink bundle-name authenticated
voice-card 0
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service h450.2
no supplementary-service h450.3
supplementary-service h450.12
h323
sip
registrar server expires max 600 min 60
username xxxxx password 0 xxxx
archive
log config
hidekeys
interface FastEthernet0/0
no ip address
ip nat inside
ip virtual-reassembly
speed auto
full-duplex
no keepalive
interface FastEthernet0/0.2
description ***** Voice LAN ******
encapsulation dot1Q 2
ip address 10.168.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/0.3
description ***** Data LAN ******
encapsulation dot1Q 3
ip address 10.168.0.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/0.4
description ***** Wireless LAN ******
encapsulation dot1Q 4
ip address 10.168.111.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1
description $ES_WAN$
no ip address
shutdown
duplex auto
speed auto
interface ATM0/1/0
no ip address
no atm ilmi-keepalive
dsl operating-mode auto
interface ATM0/1/0.1 point-to-point
pvc 0/35
pppoe-client dial-pool-number 1
interface Dialer0
no ip address
interface Dialer1
mtu 1492
ip address negotiated
no ip redirects
no ip unreachables
no ip proxy-arp
ip mtu 1452
ip nat outside
ip virtual-reassembly
encapsulation ppp
ip route-cache flow
ip tcp adjust-mss 1412
dialer pool 1
dialer-group 1
ppp authentication chap pap callin
ppp chap hostname xxxx
ppp chap password 0 xxxxx
ppp pap sent-username xxxx password 0 xxxxx
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 Dialer1
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip dns server
ip nat inside source list 101 interface Dialer1 overload
ip nat inside source route-map NAT_RMAP interface Dialer0 overload
access-list 101 permit ip 10.168.0.0 0.0.0.255 any
access-list 101 permit ip 10.168.111.0 0.0.0.255 any
route-map NAT_RMAP permit 10
match ip address 101
control-plane
voice-port 0/0/0
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
voice-port 0/0/1
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
voice-port 0/0/2
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
voice-port 0/0/3
supervisor disconnect dualtone mid-call
input gain 14
compand-type a-law
cptone BE
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 121
impedance complex2
caller-id alerting dsp-pre-allocate
dial-peer cor custom
name local
name longdistance
dial-peer cor list call-local
member local
dial-peer cor list call-longdistance
member longdistance
dial-peer cor list manager
member local
member longdistance
dial-peer cor list other
member local
dial-peer voice 1 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/0
dial-peer voice 2 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/1
dial-peer voice 3 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/2
dial-peer voice 4 pots
corlist outgoing call-longdistance
destination-pattern 0.T
port 0/0/3
dial-peer voice 5 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/0
dial-peer voice 6 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/1
dial-peer voice 7 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/2
dial-peer voice 8 pots
corlist outgoing call-local
destination-pattern 0[2,9].......
port 0/0/3
dial-peer voice 9 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/0
prefix 800
dial-peer voice 10 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/1
prefix 800
dial-peer voice 11 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/2
prefix 800
dial-peer voice 12 pots
corlist outgoing call-local
destination-pattern 0800T
port 0/0/3
prefix 800
telephony-service
no auto-reg-ephone
max-ephones 25
max-dn 25
ip source-address 10.168.10.1 port 2000
auto assign 1 to 20
timeouts interdigit 5
system message Home
max-conferences 8 gain -6
call-forward pattern .T
dn-webedit
transfer-system full-consult dss
transfer-pattern .T
ephone-dn 1 dual-line
number 101
label M.M Office (101)
description M.M Office (101)
name M.M Office
ephone 1
device-security-mode none
video
mac-address 1C17.D3C3.7A31
speed-dial 1 102 label "abc"
speed-dial 2 103 label "def"
speed-dial 3 104 label "hij"
speed-dial 4 113 label "klm"
speed-dial 5 104 label "nop"
speed-dial 6 105 label "qrst"
speed-dial 7 105 label "uvw"
type 7975
button 1:1 2:21
ephone-hunt 1 sequential
pilot 150
list 121
line con 0
exec-timeout 0 0
logging synchronous
login local
line aux 0
line vty 0 4
login local
transport input telnet
scheduler allocate 20000 1000
ntp master 2
ntp update-calendar
end
Router_Home#sh di
Router_Home#sh dia
% Ambiguous command: "sh dia"
Router_Home#sh ds
Router_Home#sh dsv   p
Router_Home#sh dspfarm
% Incomplete command.
Router_Home#sh dspfarm ?
all      Display all DSPFARM global info
dsp      Display DSPFARM DSPs information
profile Display DSPFARM profiles
Router_Home#sh dspfarm ds
Router_Home#sh dspfarm dsp
% Incomplete command.
Router_Home#sh dspfarm dsp             all
Total number of DSPFARM DSP channel(s) 0
Router_Home#sh ver
Router_Home#sh version
Cisco IOS Software, 2801 Software (C2801-ADVIPSERVICESK9-M), Version 12.4(15)T10, RELEASE SOFTWARE (fc3)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2009 by Cisco Systems, Inc.
Compiled Mon 14-Sep-09 14:51 by prod_rel_team
ROM: System Bootstrap, Version 12.3(8r)T9, RELEASE SOFTWARE (fc1)
Router_Home uptime is 21 hours, 11 minutes
System returned to ROM by power-on
System restarted at 17:31:27 UTC Wed Feb 2 2011
System image file is "flash:c2801-advipservicesk9-mz.124-15.T10.bin"
This product contains cryptographic features and is subject to United
States and local country laws governing import, export, transfer and
use. Delivery of Cisco cryptographic products does not imply
third-party authority to import, export, distribute or use encryption.
Importers, exporters, distributors and users are responsible for
compliance with U.S. and local country laws. By using this product you
agree to comply with applicable laws and regulations. If you are unable
to comply with U.S. and local laws, return this product immediately.
A summary of U.S. laws governing Cisco cryptographic products may be found at:
http://www.cisco.com/wwl/export/crypto/tool/stqrg.html
If you require further assistance please contact us by sending email to
[email protected].
Cisco 2801 (revision 6.0) with 237568K/24576K bytes of memory.
Processor board ID FCZ10491208
2 FastEthernet interfaces
1 ATM interface
2 Virtual Private Network (VPN) Modules
4 Voice FXO interfaces
1 DSP, 8 Voice resources
DRAM configuration is 64 bits wide with parity disabled.
191K bytes of NVRAM.
62720K bytes of ATA CompactFlash (Read/Write)
Configuration register is 0x2102
Router_Home#exit
Regards

Actually this has been answered already:
https://supportforums.cisco.com/thread/2148571?tstart=0
please do not open duplicate threads. You can delete you post using the Actions panel on the right.

3825 FXO Port remains in off-hook after call

Hello,
I have a 3825 router with 8 FXO ports running Cisco IOS Software, 3800 Software (C3825-SPSERVICESK9-M), Version 12.4(24)T3, RELEASE SOFTWARE (fc2). The problem we are facing is that after a call is placed through any of the FXO ports and the call is ended by the user, the port remains in off-hook till a reset of the port is done or someone restarts the router. Only then is the port accessible again.
I am thinking of changing the cards, but i do not want to invest in replacing the cards and then find out that this doesnt solve the problem.
The wierd thing is that this issue started on its own accord not too long ago.
Comments and suggestions please!
Regards,
Femi

Hello,
I do not want to change the FXO card till I am sure that is the problem and I did state that I always have to reboot the router when the problem starts. Rebooting clears the problem but it is back immediately I attempt a call again and hang up that call.
I have timeouts call-disconnect already configured, see below:
voice-port 0/0/0
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/0/1
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 2626878
caller-id enable type 1
voice-port 0/0/2
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/0/3
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/0
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/1
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/2
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
voice-port 0/1/3
supervisory disconnect dualtone mid-call
compand-type a-law
timeouts call-disconnect 5
timeouts wait-release 5
connection plar opx 21000
description FXO CONNECTION TO PSTN
caller-id enable
Regards,
Femi

CME - Sending outbound calls to FXO port

Hi Guys,
Need your help for the below scenario.
Our customer has a CME where 4 FXO ports are already connected and working. Customer has added 2 more FXO port and few IP phones.
The requirement is when ever an outbound call is made from the newly configured IP phones, the call should go through the newly added FXO lines.
For eg ext 3001 , the outbound call should go through port 0/1/0
Already the prefix 9 is used for dialing the number and I guess only one prefix number can be used in CME.
I tried translation rule , cor list but none worked , the call is default going through the old fxo port and not to the new fxo port.
Can you guys help me with the configuration.
Regards
Sathya

Previous post on similar issue might be helpful -
https://supportforums.cisco.com/discussion/11431746/h323-choose-outbound-fxo-port-based-calling-number
Thnx

Uc560 fxo port not answering incoming calls

Hi,
My customer is facing problem for incoming calls in uc560 fxo port.They have 12 PSTN lines which is connected to UC system.System is configured with Auto-Attendant also. almost all days they are facing this major issue of incoming call is not getting answered by UC560 and caller can hear the line is ringing.While the time of this problem I can see some of the FXO port status LED is UP and not disconnecting even if no one is on call also.Once remove the cable from the FXO port and connect it back the problem will solve for time being.What will be the reason for this issue of line getting held.Is there any configuration needs to change in FXO module? Below is the configuration I done on all 12 FXO ports. Please check and
suggest me a solution.

HI Paolo,
Thank you provoding the proper documentation .
On the system side I made the change by keeping companding type from a-law to u-law and enabled battery reverse.This setting works fine for last three days and now again the customer is facing the same problem of FXO port get held and incoming calls are just ringing and system is not answering even.
How to get proper solution for this issue????
Please help me............
Regards,
Rinchuraj

Call disconnect on FXO port

Hi All,
I have an issue with call disconnect on FXO port. Call are getting disconnected as soon as it connects.
I observed the call takes the port 0/20 which has the command supervisory disconnect anytone
but in other ports I see supervisory disconnect dualtone mid-call
so I have changed with the second and got it working for some calls, In the debug vpm singal debugs I see remote device releases the call.
Could you tell me how this works and what are the debugs we could check to see what is the exact problem?
Number dialled are : 361000, 352788
Can you check? Logs and sh run is attached.
Thanks,
Lajith P

Beside that what you said Laith - I still think that you need specific tones for disconnect on FXO and only then your voice setup is going to work as it should. I can't comment on different version of IOS-es but maybe your previous one was better in handling this type of problem...if you don't have any specific requirement and must use 15.1 IOS you can always revert back to 15.0 and continue using your ports as already...
But if I was on your place - I would solve this problem once for all and then you don't have to worry about some upcoming new IOS
HTH,
Dragan

Debug infor about FXO port

I have a debug information showed by command"debug vpm all"about FXO port.I can't distinguish the error in the information. who can tell me how to find out the error?
Thanks.
debug infor:
3d18h: ccIFShowState (vdbPtr=0x62F47D14, summary)
3d18h: ccIFShowState (vdbPtr=0x62F49994, summary)
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x0 timestamp=48919 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_rig
3d18h: [3/0/0] htsp_start_caller_id_rx
3d18h: [3/0/0] htsp_set_caller_id_rx:BELLCORE
3d18h: htsp_timer - 125 msec
3d18h: htsp_process_event: [3/0/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxolr
3d18h: htsp_timer - 10000 msec
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=49519 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
3d18h: fxols_ringing_not
3d18h: htsp_timer_stop
3d18h: htsp_timer - 10000 msec
3d18h: [3/0/0] htsp_stop_caller_id_rx
3d18h: hdsprm_close_cleanup
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x0 timestamp=51819 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_RINGING, E_DSP_SIG_0000]
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=52819 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_RINGING, E_DSP_SIG_0100]
3d18h: fxols_ringing_not
3d18h: htsp_timer_stop htsp_setup_ind
3d18h: [3/0/0] get_fxo_caller_id:Caller ID received. Message type=4 length=18 c3
3d18h: [3/0/0] Caller ID String 04 0F 30 37 31 38 31 35 35 31 35 32 36 33 39 33
3d18h: [3/0/0] get_fxo_caller_id calling num=5263932 calling name= calling time
3d18h: cc_api_call_setup_ind (vdbPtr=0x62F47D14, callInfo={called=8059,called_o)
3d18h: cc_api_call_setup_ind type 2 , prot 0
3d18h: cc_insert_call_entry: Increment call volume counter
3d18h: cc_insert_call_entry: current call volume: 1
3d18h: cc_insert_call_entry: entry's incoming TRUE. is_incoming is TRUE
3d18h: cc_incr_if_call_volume: not the VoIP or MMoIP
3d18h: htsp_process_event: [3/0/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
3d18h: fxols_wait_setup_ack:
3d18h: [3/0/0] set signal state = 0xC timestamp = 0
3d18h: dsp_set_sig_state: [3/0/0] packet_len=12 channel_id=128 packet_id=39 sta0
3d18h: dsp_soutput: [3/0/0]fxols_check_auto_call
3d18h: cc_process_call_setup_ind (event=0x62ED923C)
3d18h: >>>>CCAPI handed cid 25 with tag 200 to app "DEFAULT"
3d18h: sess_appl: ev(24=CC_EV_CALL_SETUP_IND), cid(25), disp(0)
3d18h: sess_appl: ev(SSA_EV_CALL_SETUP_IND), cid(25), disp(0)
3d18h: ssaCallSetupInd
3d18h: ccCallSetContext (callID=0x19, context=0x631FB148)
3d18h: ssaCallSetupInd cid(25), st(SSA_CS_MAPPING),oldst(0), ev(24)ev->e.evCall1
3d18h: ssaCallSetupInd finalDest cllng(5263932), clled(8059)
3d18h: ssaCallSetupInd cid(25), st(SSA_CS_CALL_SETTING),oldst(0), ev(24)dpMatch0
3d18h: ssaSetupPeer cid(25) peer list: tag(100) called number (8059)
3d18h: ssaSetupPeer cid(25), destPat(8059), matched(4), prefix(), peer(62A8DF54)
3d18h: ccCallProceeding (callID=0x19, prog_ind=0x0)
3d18h: ccCallSetupRequest (Inbound call = 0x19, outbound peer =100, dest=, para1
3d18h: ccCallSetupRequest numbering_type 0x81
3d18h: ccCallSetupRequest encapType 2 clid_restrict_disable 1 null_orig_clg 0 c2
3d18h: dest pattern 8059, called 8059, digit_strip 0
3d18h: callingNumber=5263932, calledNumber=8059, redirectNumber= display_info= 0
3d18h: accountNumber=, finalDestFlag=1,
guid=7ea5.51a9.17e5.11cc.8034.e670.5153.4d65
3d18h: peer_tag=100
3d18h: ccIFCallSetupRequestPrivate: (vdbPtr=0x62CDA89C, dest=, callParams={call1
3d18h: ccIFCallSetupRequestPrivate: (vdbPtr=0x62CDA89C, dest=, callParams={call)
3d18h: cc_insert_call_entry: not incoming entry
3d18h: cc_insert_call_entry: entry's incoming FALSE. is_incoming is FALSE
3d18h: ccSaveDialpeerTag (callID=0x19, dialpeer_tag=0x64)
3d18h: ccCallSetContext (callID=0x1A, context=0x631FB6BC)
3d18h: ccCallReportDigits (callID=0x19, enable=0x0)
3d18h: cc_api_call_report_digits_done (vdbPtr=0x62F47D14, callID=0x19, disp=0)
3d18h: sess_appl: ev(52=CC_EV_CALL_REPORT_DIGITS_DONE), cid(25), disp(0)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(1)fDest(1)
3d18h: -cid2(26)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_MAPPING)
3d18h: ssaReportDigitsDone cid(25) peer list: (empty)
3d18h: ssaReportDigitsDone callid=25 Reporting disabled.
3d18h: htsp_process_event: [3/0/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_ofc
3d18h: htsp_timer - 120000 msec
3d18h: cc_api_supported_data data_mode=0x10002
3d18h: cc_incr_if_call_volume: remote IP is x.x.x.x
3d18h: cc_incr_if_call_volume: hwidb is Serial1/0:0
3d18h: cc_incr_if_call_volume: create entry in list: 1
3d18h: ccTDUtilGetInstanceCount: For tagID[1] of callID[26]
3d18h: ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[2]
3d18h: ccTDUtilGetInstanceCount: For tagID[2] of callID[26]
3d18h: ccTDPvtProfileTableObjectAccessManager: No profileTable set for callID[2]
3d18h: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0xC timestamp=53222 systim8
3d18h: htsp_process_event: [3/0/0, FXOLS_PROCEEDING, E_DSP_SIG_1100]fxols_offhoc
3d18h: htsp_timer2 - 350 msec
3d18h: cc_api_call_proceeding(vdbPtr=0x62CDA89C, callID=0x1A,
prog_ind=0x0, rawmsgPtr=0x0)
3d18h: sess_appl: ev(21=CC_EV_CALL_PROCEEDING), cid(26), disp(0)
3d18h: cid(26)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_PROCEEDING)
oldst(SSA_CS_MAPPING)cfid(-1)csize(0)in(0)fDest(0)
3d18h: -cid2(25)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_CALL_SETTING)
3d18h: ssaCallProc
3d18h: ccGetDialpeerTag (callID=0x19)
3d18h: ssaIgnore cid(26), st(SSA_CS_CALL_SETTING),oldst(1), ev(21)
3d18h: cc_api_call_alert(vdbPtr=0x62CDA89C, callID=0x1A, prog_ind=0x0, sig_ind=)
3d18h: sess_appl: ev(7=CC_EV_CALL_ALERT), cid(26), disp(0)
3d18h: cid(26)st(SSA_CS_CALL_SETTING)ev(SSA_EV_CALL_ALERT)
oldst(SSA_CS_CALL_SETTING)cfid(-1)csize(0)in(0)fDest(0)
3d18h: -cid2(25)st2(SSA_CS_CALL_SETTING)oldst2(SSA_CS_CALL_SETTING)
3d18h: ssaAlert
3d18h: ccGetDialpeerTag (callID=0x19)
3d18h: ccCallAlert (callID=0x19, prog_ind=0x0, sig_ind=0x1)htsp_alert_notify
3d18h: htsp_process_event: [3/0/0, FXOLS_PROCEEDING, E_HTSP_EVENT_TIMER2]fxols_m
3d18h: htsp_timer_stop
3d18h: htsp_timer_stop2
3d18h: cc_api_call_disconnected(vdbPtr=0x62F47D14, callID=0x19, cause=0x10)
3d18h: sess_appl: ev(11=CC_EV_CALL_DISCONNECTED), cid(25), disp(0)
3d18h: cid(25)st(SSA_CS_ALERT_RCVD)ev(SSA_EV_CALL_DISCONNECTED)
oldst(SSA_CS_CALL_SETTING)cfid(-1)csize(0)in(1)fDest(1)
3d18h: -cid2(26)st2(SSA_CS_ALERT_RCVD)oldst2(SSA_CS_CALL_SETTING)
3d18h: ssaDiscSetting
3d18h: ssaFlushPeerTagQueue cid(25) peer list: (empty)
3d18h: ssa: Disconnected cid(25) state(17) cause(0x10)
3d18h: ccCallDisconnect (callID=0x19, cause=0x10 tag=0x0)
3d18h: cc_api_get_transfer_info: (callID=0x19)
3d18h: ccCallDisconnect (callID=0x1A, cause=0x10 tag=0x0)
bazhong#sh voice call su
PORT CODEC VAD VTSP STATE VPM STATE
============ ======== === ==================== ======================
3/0/0 None y S_SETUP_IND_PEND FXOLS_RINGING
3/0/1 - - - FXOLS_ONHOOK
Sincerely

I just received the response from customer engineer about the issue that the problem is the IVR cannot make the correct response to the call from the branch office. the originated caller only hear the normal ring, but can't hear the IVR replied sound just like" welcome to xxx company". Could u tell me how to make the right troubleshooting to solve the problem? I have a question that whether the call can arrive the network gateway correctly, but can't be transferred to the call manager gateway correctly. Wait for your help. Thanks
Sincerely

FXO disconnect Issue

I have a problem regarding the termination of FXO ports.
We have a call manager with IP Phones connected and a Cisco 2821
gateway with FXO ports.Now the issue is that when somebody calls
from PSTN to the IP Phone and disconnects the call before the IP PHone
user pick up the phone ,the IP PHone keeps ringing and the FXO port remains
up though the PSTN user has already disconnected the call.
I have read many documents regarding this issue on Cisco Web site.We successfully
resolved the issue but another problem arised after that.
NOw here are the 3 steps we did.
1)INitially we did not apply any configuration to the FXO ports .So we
were having the same problem as described above.
2)We applied the following configuration on the FXO Ports,
voice class custom-cptone 1
dualtone busy
dualtone ringback
dualtone reorder
dualtone out-of-service
dualtone number-unobtainable
dualtone disconnect
frequency 425 425
cadence 200 200 200 200
voice-port 0/1/0
supervisory disconnect anytone
supervisory custom-cptone 1
input gain -6
output attenuation 6
echo-cancel coverage 24
cptone SA
caller-id enable
The problem resolved succesfully .The FXO port goes on hook when the
PSTN user disconnects the call.
3)After applying the above config,we used a tcl script to provide
basic auto-attendant feature on the router.We started having the same FXO issue
again.Now based upon the above information ,how do u think we should
resolve the problem.

This URL should help you:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml

Calls out FXO port fail

I can't make any calls out this FXO port. Any ideas? Is it a telco issue?
IOS is 124-24.T4.bin
.Nov 18 06:12:29: htsp_timer_stop3
.Nov 18 06:12:29: [0/1/1] htsp_stop_caller_id_rx. message length 0htsp_setup_req
.Nov 18 06:12:29: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_HTSP_SETUP_REQ]fxols_onhook_setup
.Nov 18 06:12:29: [0/1/1] set signal state = 0xC timestamp = 0
.Nov 18 06:12:29: htsp_timer - 1300 msec
.Nov 18 06:12:29: [0/1/1] htsp_dsm_close_done
.Nov 18 06:12:30: htsp_process_event: [0/1/1, FXOLS_WAIT_DIAL_TONE, E_DSP_SIG_0110]fxols_disc_clear
.Nov 18 06:12:30: htsp_timer_stop2
.Nov 18 06:12:30: htsp_timer - 1300 msec
.Nov 18 06:12:31: htsp_process_event: [0/1/1, FXOLS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER]fxols_wait_dial_timer htsp_dial
.Nov 18 06:12:33: htsp_process_event: [0/1/1, FXOLS_WAIT_DIAL_DONE, E_DSP_DIALING_DONE]fxols_wait_dial_done htsp_progress
.Nov 18 06:12:33: htsp_timer - 350 msec
.Nov 18 06:12:33: htsp_call_bridged invoked
.Nov 18 06:12:33: htsp_process_event: [0/1/1, FXOLS_WAIT_CUT_THRU, E_HTSP_VOICE_CUT_THROUGH]fxols_handle_cut_thru
.Nov 18 06:12:33: htsp_timer_stop
.Nov 18 06:12:34: htsp_process_event: [0/1/1, FXOLS_OFFHOOK, E_DSP_SUP_DISCONNECT]fxols_outgoing_sup_disc
.Nov 18 06:12:34: htsp_timer - 3000 msec
.Nov 18 06:12:37: htsp_process_event: [0/1/1, FXOLS_OFFHOOK, E_HTSP_EVENT_TIMER]fxols_disc_confirm
.Nov 18 06:12:37: htsp_timer_stop
.Nov 18 06:12:37: htsp_timer_stop2
.Nov 18 06:12:37: htsp_timer_stop3
.Nov 18 06:12:37: htsp_timer_stop3
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2, ConnectionId 3FE4209F37B14654000008A43C0E8F, SetupTime .06:12:29.778 GTM Tue Nov 18 2014, PeerAddress 605281980020, PeerSubAddress , DisconnectCause 11 , DisconnectText user busy (17), ConnectTime .06:12:33.158 GTM Tue Nov 18 2014, DisconnectTime .06:12:37.298 GTM Tue Nov 18 2014, CallOrigin 2, ChargedUnits 0, InfoType 2, TransmitPackets 102, TransmitBytes 4080, ReceivePackets 203, ReceiveBytes 4060
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:11/18/2014 06:12:29.782,cgn:605281980020,cdn:080006099,frs:0,fid:45246,fcid:3FE4209F37B14654000008A43C0E8F,legID:10EF5,bguid:003FE4209F37B14654000008A43C0E8F
.Nov 18 06:12:37: htsp_process_event: [0/1/1, FXOLS_REMOTE_RELEASE, E_HTSP_RELEASE_REQ]fxols_offhook_release
.Nov 18 06:12:37: htsp_timer_stop
.Nov 18 06:12:37: htsp_timer_stop2
.Nov 18 06:12:37: htsp_timer_stop3
.Nov 18 06:12:37: [0/1/1] set signal state = 0x4 timestamp = 0
.Nov 18 06:12:37: htsp_timer - 2000 msec
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1, ConnectionId 3FE4209F37B14654000008A43C0E8F, SetupTime .06:12:29.790 GTM Tue Nov 18 2014, PeerAddress 080006099, PeerSubAddress , DisconnectCause 11 , DisconnectText user busy (17), ConnectTime .06:12:33.160 GTM Tue Nov 18 2014, DisconnectTime .06:12:37.280 GTM Tue Nov 18 2014, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 102, TransmitBytes 4080, ReceivePackets 203, ReceiveBytes 5684
.Nov 18 06:12:37: %VOIPAAA-5-VOIP_FEAT_HISTORY: FEAT_VSA=fn:TWC,ft:11/18/2014 06:12:29.786,cgn:605281980020,cdn:080006099,frs:0,fid:45247,fcid:3FE4209F37B14654000008A43C0E8F,legID:10EF6,bguid:003FE4209F37B14654000008A43C0E8F
.Nov 18 06:12:37: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
.Nov 18 06:12:39: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
.Nov 18 06:12:39: fxols_dsp_prealloc_clid_wait. Line reversal alerting DSP preallocation done
.Nov 18 06:12:39: [0/1/1] htsp_start_caller_id_rx:BELLCORE
.Nov 18 06:12:39: htsp_start_caller_id_rx create dsp_stream_manager
.Nov 18 06:12:39: [0/1/1] htsp_dsm_create_success returns 1
.Nov 18 06:12:39: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
voice-port 0/1/1
supervisory disconnect dualtone mid-call
cptone MX
timeouts interdigit 4
timeouts call-disconnect 3
timeouts wait-release 3
caller-id enable
caller-id alerting dsp-pre-allocate

Hi Jason,
Have you tried disabling the command "supervisory disconnect dualtone mid-call"?
Did you have DSP installed?
Regards

Line polarity important for FXO ports

I spent several hours trying to get the FXO ports working on my UC320W without success until, in a moment of desperation, I reversed the polarity on the phone line I was using for testing and suddenly everything's working nicely. I've gotten spoiled with other equipment that automatically adjusts to reversed polarity - apparently that's not true for the UC320 or the 8800 units.
Just thought maybe I'd save someone else some aggravation, as I saw no mention of this as a troubleshooting step in the various materials available.

Hi Greg,
Thank you for your comments. Couple of additional questions.
Are you located in North America? What type of issues were you having when the polarity was reversed? i.e Call ID or FXO ports not answering, not able to make outbound call... etc..Your feedback is appreciated. I'm checking with our hardware engineers on the issue you raised. I'm also confirming that FCC Part 68 complaince tests for fxo reverse polarity conditions. If your available I'd like to reach out to you after I receive feedback from our hardware team.
Regards,
Randy

FXO port not going back onhook when call goes to CUC Voice mail.

HI All,
Quiet new to the voice world and have just set up a home lab and have run into the following problem.
When a call comes in from the PSTN and gets transferred to voicemail CUC, when the caller has left the message and hung up the FXO port on my Cisco 3825 does not go back on hook
If a call comes in from the PSTN and is answered on a phone registered to the CUCM and the internal user hangs up the FXO port goes back on hook but if the caller hangs up the FXO port does not go back on hook.
I am using H.323 between the 3825 & CUCM
PSTN ------->Voice Gateway 3825-------->CUCM-------->CUC
Anyone got any ideas as to why this is happening?
Thanks

Hi Shaun,
The router DSP port is not able to detect the disconnect tone sent by provider that is why FXO port is remaining off hook. When call is answered by IP phone then IP phone user can listen the disconnect tone when PSTN caller disconnectes the call and they will hang up the call so gateway is aware that call is disconnected and moves the port to ON hook state but CUE cannot understand the tone so it keeps the call active. To resolve the issue we ave to configure custom CPTONE. Kindly collect PCM capture for a test call. Disable capture after call is disconnected by PSTN caller. i have attached the document. Use the method according to your IOS version.
Using PCM capture we can findout the exact frequency and cadence of the disconnect tone sent by provider.

Any known 'porting' issues between 8.2.2 and 9.0 ??

Hi all,
we are in the process of evaluating LS ES2 (9.0) and would like to know if there are any 'porting' issues between the two versions. Asked another way, are there any issues in deploying a 8.2.2. LCA onto 9.0 ?
Dan

8.2 LCA's will run fine in 9.0 if they are just deployed into it. But in 9.0 a new application model has been introduced, so in order to make changes to anything after you've migrated it you'll need to create an application and migrate your processes and resources into it. You may want to take a look at the Leveraging Legacy Solution document available online. Here's the link: http://help.adobe.com/en_US/livecycle/9.0/workbenchHelp/help.htm?content=leveragingLegacy. html .
Chris

FXO port issue

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