FXS FXO plar problem

I have VOIP configured with FXO and FXS,now I can plar from FXO to FXS but when I plar from FXS to FXO I can call in a few second and it will disconnect line.please sugguest me.
FXS config
call rsvp-sync
voice-port 2/0
input gain 8
description 8001
voice-port 2/1
input gain 8
description 8002
mgcp profile default
dial-peer cor custom
dial-peer voice 8001 pots
destination-pattern 8001
port 2/0
dial-peer voice 8002 pots
destination-pattern 8002
port 2/1
dial-peer voice 7001 voip
destination-pattern 900.
session target ipv4:10.10.11.2
codec g729r8 bytes 40
ip qos dscp cs5 media
FXO Config
call rsvp-sync
voice-port 2/0
input gain 8
connection plar 8001
voice-port 2/1
input gain 8
connection plar 8002
mgcp profile default
dial-peer cor custom
dial-peer voice 9001 pots
destination-pattern 9001
port 2/0
dial-peer voice 9002 pots
destination-pattern 9002
port 2/1
dial-peer voice 9003 voip
destination-pattern 800.
session target ipv4:10.10.11.1
codec g729r8 bytes 40
ip qos dscp cs5 media
When I debug voip ccapi error I see this message
*Mar 1 04:30:10.561: //211/8E60753E81AC/SSAPP:9003:93/ssaConfCreateDoneAlert: O
ther call leg not found
*Mar 1 04:31:41.021: //211/8E60753E81AC/CCAPI/cc_api_call_disconnect_done: caus
e=16,retry=0,vcCauseCode=0
*Mar 1 04:31:41.021: //212/8E60753E81AC/CCAPI/ccGetCallActiveByCallID: cc_spi_c
all_get() returned -7. (setup_time=0x18BC3F, index=0x1)
*Mar 1 04:31:41.029: //212/8E60753E81AC/CCAPI/cc_api_call_disconnect_done: caus
e=16,retry=0,vcCauseCode=0
*Mar 1 04:31:55.806: //212/8E60753E81AC/SSAPP:9003:93/ssaConfCreateDoneAlert: O
ther call leg not found

before I config that you suggest,I try to test by connect the phone direct to FXS port.It can use plar
and call doesn't disconnect.It will have problem with plar when FXS use PBX.What is parameter to tuning?How to check it?
Thank you.

Similar Messages

FXO to FXO disconnect problem - Part1

Hi,
I have 2 sites connected through an hdlc leased line. Each site has router 1750 with one VIC - 2FXO.
Once a call is placed both FXO are stuck in the off hook state. So there is no way to make another call unless I shut down and turn on the ViC again.
I read the "famous" fxo disconnect problem file from Cisco but it didn't help.
I need to know:
1) Is this problem a bug on the FXO ports ?
2) Is this problem from the analog PBX side ?
3) How to solve this problem?
Thanks

Hi,
you can use FXS cards......you don't have any problem......
Problem is in disconnect reason....FXO card no detect disconnect tone from PBX....
Czech firm 2N sell special "disconnect module" fot these card......
(I'm sorry for my English...:o)

FXO Disconnect Problem

Hello,
I just would like to ask some ideas on FXO disconnect problem having a setup of FXO to FXO termination on both sites. I am using a VIC-2FXO-M2 on both Cisco 1760-V Router and these fxo ports are terminated to the PABX local.The voice port will disconnect 30minutes (approximately) after the call is terminated or it remainded off-hook although the call is already done.
We tried some fine tunings like enabling "battery-reversal" command, setting the timeout call disconnect to 5sec, timeout wait-release to 5sec. But the problem is still the same.
What is really the ideal fxo disconnect solution that we can do to lessen the 30minutes delay from off-hook to on-hook.
Looking forward for your great ideas.
Thank you.
Vivira Alastra

Dear Vivira,
The disconnection issue of FXO ports can really produce some gray hairs...
I have worked integrated to some cheap Panasonic PBX's in Nigeria where documentation is really scarce.
The solution that worked - but don't ask me how - was to try different combinations of CPTone on the Voice Ports. That did the job in our case.
You can also try the tedious job of measuring the disconnect tones ans analyse it e.g. in CoolEdit and then set up a Voice Class where you define the tones.
But try the CPTone settings first.
Good luck.
Peter

Cisco Solution for FXO Disconnect Problem

I have Panasonic PBX-->FXO-->1751V------WAN-----1751V-->FXO--Panasonic PBX. I am facing a serious problem with FXO on both ends becoming busy after some few initial calls.
What is the exact Cisco solution for this FXO disconnect problem? I have read and applied some Netpro tips for similar case but none could help.
Can we have Cisco expert explain this??

Hello, I think this url may be helpful:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml
Regards.

FXO to FXO disconnection problem -- Help me please.

My environment is as follows :
phoneA--<PBX>-FXO-<Router2821>--WAN--<Router2801>-FXO-<PBX>-phoneB
PhoneA NO# is 375
PhoneB NO# is 312
the related configurations from Router2821 are as follows:
voice-port 1/1/3
supervisory disconnect anytone
cptone TH
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
connection plar 614
dial-peer voice 8 pots
destination-pattern 379
port 1/1/3
the related configurations form Router2801 are as follows:
voice-port 0/0/3
supervisory disconnect anytone
cptone TH
timeouts call-disconnect 5
timeouts ringing 10
timeouts wait-release 1
connection plar 379
dial-peer voice 4 pots
destination-pattern 614
port 0/0/3
Steps to reproduce to problem are as follows:
1. Phone A dials to 379 ( the assigned NO on FXO of 2821 )
then the connection goes to 614( the assigned NO on FXO of 2801) by plar feature.
2. After got a tone from FXO of 2801, PhoneA dials to NO# 312( Phone B NO#)
and then PhoneB starts ringing.
3. While PhoneB is ringing( NoOne accepts the call ) PhoneA hooks-on the Phone.
--- The problem is PhoneB remains ringing and the FXO of both routers are not released ---
From the configuration above I have applied ringing timeout,disconnect timeout and wait-release timeout
to both routers. But it can't solve the problem.
Could anyone help me to solve this issue ?
Thanks a lot.

I also experienced same problem. I have diagram
- - RouterA <---> RouterB - -
when i put a call from phoneA to phoneB, the same problem occurs.
detail:
phoneA : 101
routerA : 1760
routerB : 1760
phoneB : 630
I have followed the procedure from linksite that had been posted. but the matter still occur. the information and result of debug
RouterB# sh voice call summ
PORT CODEC VAD VTSP STATE VPM STATE
2/0 g729r8 y S_CONNECT FXOLS_OFFHOOK
2/1 - - - FOLS_ONHOOK
(debug routerB)
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_ONHOOK, E_HTSP_SETUP_REQ]fxogs_onhook_setup
Dec 23 00:08:26: [3/1] set signal state = 0x0 timestamp = 0
Dec 23 00:08:26: dsp_set_sig_state: [3/1] packet_len=12 channel_id=129 packet_id=39 state=0x0 timestamp=0x0
Dec 23 00:08:26: TGRM: reg_invoke_tgrm_call_update(0, 3, 1, 0, 1, TGRM_CALL_BUSY, TGRM_CALL_VOICE, TGRM_DIRECTION_OUT)
Dec 23 00:08:26: htsp_timer - 10000 msec
Dec 23 00:08:26: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=56945 systime=639197437
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_WAIT_TIP_GROUND, E_DSP_SIG_0100]fxogs_start_dial
Dec 23 00:08:26: htsp_timer_stop
Dec 23 00:08:26: [3/1] set signal state = 0xC timestamp = 0
Dec 23 00:08:26: dsp_set_sig_state: [3/1] packet_len=12 channel_id=129 packet_id=39 state=0xC timestamp=0x0
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_WAIT_DIAL_TONE, E_HTSP_EVENT_TIMER]fxogs_wait_dial_timer htsp_dial
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_WAIT_DIAL_DONE, E_DSP_DIALING_DONE]fxogs_wait_dial_donehtsp_connect: no_offhook 0htsp_progress
Dec 23 00:08:26: htsp_timer - 350 msec
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_WAIT_ONHOOK, E_HTSP_CONNECT]
Dec 23 00:08:26: [3/1, FXOGS_WAIT_ONHOOK, E_HTSP_CONNECT] -> ERROR: INVALID INPUT
Dec 23 00:08:26: ipm_modem_relay_supported : false
Dec 23 00:08:26: ipm_modem_relay_supported : false
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_WAIT_ONHOOK, E_HTSP_VOICE_CUT_THROUGH]fxogs_handle_cut_thru
Dec 23 00:08:26: htsp_timer_stop
Dec 23 00:08:26: dsp_req_sig_state: [3/1] packet_len=8 channel_id=129 packet_id=40
Dec 23 00:08:26: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=56945 systime=639197438
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_OFFHOOK, E_DSP_SIG_0100]
Dec 23 00:08:26: fxogs_stop_disc_timer
Dec 23 00:08:26: htsp_timer_stop2
Dec 23 00:08:26: htsp_dsp_message: SEND/RESP_SIG_STATUS: state=0x4 timestamp=57228 systime=639197465
Dec 23 00:08:26: htsp_process_event: [3/1, FXOGS_OFFHOOK, E_DSP_SIG_0100]
Dec 23 00:08:26: fxogs_stop_disc_timer
Dec 23 00:08:26: htsp_timer_stop2
note:
both router use IOS 12.3.17a, dspware 4.1.41
and the router doesn't have command for tip_ground
please help..

EVM module FXS - FXO GSM Gateway interoperability issue

Hello,
I have a router Cisco 2821 with EVM module 8 FXS + 4 BRI and GSM Gatewat Ecotel Vierling GSM3-1F which has a FXO and FXS ports.
I'm controlling the router interfaces by mgcp and tried to connect the GSM Gateway. The problem is that the GSM Gateway expects the digits after it goes off-hook but the CCM sends them before that.
I want to develop a method to send the digits inband.
Now it works with access code 8 and then users dial the phone when they head the second dial tone but it is not very convenient.
Further thank for any ideas.
Best Regards,
Zdravkov

Look at this link for more information on GSM Gateway:
http://www.cisco.com/en/US/prod/collateral/iosswrel/ps8802/ps6968/ps6441/product_bulletin_c25-409474.html

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.

Cisco FXO Disconnect Problem

we have 2801 router that connected with 2 anloga lines (FXO Card), but now we have a problem with disconnect problem, the phone still connected after the PSTN Caller Disconnect, and our policy tell that the agents shouldn't end the call, so we want when the PSTN caller end the call we need the phone to return to idle state.
and according for the below link, i configure the custom disconnect tone, and in the attachment you can find tow disconnect tones.
http://ciscoflair.blogspot.com/2009/05/cisco-fxo-disconnect-issue.html
the configuration of the custom disconnect tone
voice class dualtone-detect-params 1
cadence-variation 3
voice class custom-cptone Custom1
dualtone disconnect
frequency 420
cadence 245 255 245 255 245 255
and the below you can find the configuration of the voice port, and you can find the debug vpm port 0/3/0, debug vpm signal, for a call that disconnected immediately, and for a call the take long time to disconnected.
this is the configuration of the voice port:
voice-port 0/3/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone Custom1
supervisory dualtone-detect-params 1
no battery-reversal
cptone NL
timeouts call-disconnect 5
timeouts wait-release 5
timing hookflash-out 50
timing guard-out 300
caller-id enable
caller-id alerting line-reversal
caller-id alerting dsp-pre-allocate
this is the debug of a call that is disconnected immediately:
Jan 25 11:48:32.262: [0/3/0] htsp_dsm_feature_notify_cb returns 2 id=DSM_FEATURE_SM_CALLERID_RX
Jan 25 11:48:32.262: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_CALLERID_RX_DONE]
Jan 25 11:48:32.262: [0/3/0] htsp_stop_caller_id_rx. message length 11
Jan 25 11:48:32.262: [0/3/0] htsp_dsm_close_done
Jan 25 11:48:34.026: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Jan 25 11:48:34.026: htsp_timer - 125 msec
Jan 25 11:48:34.154: htsp_process_event: [0/3/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Jan 25 11:48:34.154: htsp_timer - 10000 msec
Jan 25 11:48:35.305: htsp_process_event: [0/3/0, FXOLS_RINGING, E_DSP_SIG_0100]
Jan 25 11:48:35.305: fxols_ringing_not
Jan 25 11:48:35.305: htsp_timer_stop
Jan 25 11:48:35.305: htsp_timer_stop3 htsp_setup_ind
Jan 25 11:48:35.305: [0/3/0] get_fxo_caller_id:Caller ID received. Message type=129 length=11 checksum=00
Jan 25 11:48:35.309: [0/3/0] Caller ID String 44 30 36 35 36 37 39 31 34 31
Jan 25 11:48:35.309: [0/3/0] get_fxo_caller_id calling num=065679141 calling name= calling time=01/25 13:48
Jan 25 11:48:35.313: htsp_process_event: [0/3/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Jan 25 11:48:35.313: fxols_wait_setup_ack:
Jan 25 11:48:35.313: [0/3/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Jan 25 11:48:35.321: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Jan 25 11:48:35.321: htsp_timer - 120000 msechtsp_alert_notify
Jan 25 11:48:35.389: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alerthtsp_call_service_msghtsp_call_service_msg not EFXS (2)
Jan 25 11:48:35.509: htsp_call_bridged invokedhtsp_call_service_msghtsp_call_service_msg not EFXS (2)
Jan 25 11:48:35.513: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Jan 25 11:48:35.513: htsp_timer_stop
Jan 25 11:48:35.521: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Jan 25 11:48:35.585: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Jan 25 11:48:35.585: htsp_timer_stop2
sh voice po su
                                           IN       OUT
PORT            CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
=============== == ============ ===== ==== ======== ======== ==
0/3/0           -- fxo-ls      up    up   idle     off-hook y
0/3/1           -- fxo-ls      up    dorm idle     on-hook y
50/0/1          1      efxs     up    up   on-hook idle     y
50/0/1          2      efxs     up    up   on-hook idle     y
PWR FAILOVER PORT        PSTN FAILOVER PORT
=================        ==================
Jan 25 11:48:43.857: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_DSP_SUP_DISCONNECT]fxols_conn_sup_disc
Jan 25 11:48:43.857: htsp_timer2 - 5000 msec
Madaba_Maint#
Jan 25 11:48:48.856: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_EVENT_TIMER2]fxols_disc_confirm
Jan 25 11:48:48.856: htsp_timer_stop
Jan 25 11:48:48.856: htsp_timer_stop2
Jan 25 11:48:48.856: htsp_timer_stop3
Jan 25 11:48:48.860: htsp_timer_stop3
Jan 25 11:48:48.876: htsp_process_event: [0/3/0, FXOLS_REMOTE_RELEASE, E_HTSP_RELEASE_REQ]fxols_offhook_release
Jan 25 11:48:48.876: htsp_timer_stop
Jan 25 11:48:48.876: htsp_timer_stop2
Jan 25 11:48:48.876: htsp_timer_stop3
Jan 25 11:48:48.876: [0/3/0] set signal state = 0x4 timestamp = 0
Jan 25 11:48:48.876: htsp_timer - 300 msec
Jan 25 11:48:49.148: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Jan 25 11:48:49.176: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Jan 25 11:48:49.176: fxols_dsp_prealloc_clid_wait. Line reversal alerting DSP preallocation done
Jan 25 11:48:49.176: [0/3/0] htsp_start_caller_id_rx:ETSI_DTMF
Jan 25 11:48:49.176: htsp_start_caller_id_rx create dsp_stream_manager
Jan 25 11:48:49.176: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jan 25 11:48:49.176: [0/3/0] htsp_dsm_create_success returns 1
Madaba_Maint#sh voice po su
                                           IN       OUT
PORT            CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
=============== == ============ ===== ==== ======== ======== ==
0/3/0           -- fxo-ls      up    dorm idle     on-hook y
0/3/1           -- fxo-ls      up    dorm idle     on-hook y
50/0/1          1      efxs     up    up   on-hook idle     y
50/0/1          2      efxs     up    up   on-hook idle     y
this is a debug for a call that is take a long time to disconnected:
Jan 25 11:49:42.267: [0/3/0] htsp_dsm_feature_notify_cb returns 2 id=DSM_FEATURE_SM_CALLERID_RX
Jan 25 11:49:42.267: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_HTSP_CALLERID_RX_DONE]
Jan 25 11:49:42.267: [0/3/0] htsp_stop_caller_id_rx. message length 11
Jan 25 11:49:42.271: [0/3/0] htsp_dsm_close_done
Jan 25 11:49:43.999: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
Jan 25 11:49:43.999: htsp_timer - 125 msec
Jan 25 11:49:44.127: htsp_process_event: [0/3/0, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
Jan 25 11:49:44.127: htsp_timer - 10000 msec
Jan 25 11:49:45.279: htsp_process_event: [0/3/0, FXOLS_RINGING, E_DSP_SIG_0100]
Jan 25 11:49:45.279: fxols_ringing_not
Jan 25 11:49:45.279: htsp_timer_stop
Jan 25 11:49:45.279: htsp_timer_stop3 htsp_setup_ind
Jan 25 11:49:45.279: [0/3/0] get_fxo_caller_id:Caller ID received. Message type=129 length=11 checksum=00
Jan 25 11:49:45.279: [0/3/0] Caller ID String 44 30 36 35 36 37 39 31 34 31
Jan 25 11:49:45.279: [0/3/0] get_fxo_caller_id calling num=065679141 calling name= calling time=01/25 13:49
Jan 25 11:49:45.283: htsp_process_event: [0/3/0, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
Jan 25 11:49:45.283: fxols_wait_setup_ack:
Jan 25 11:49:45.287: [0/3/0] set signal state = 0xC timestamp = 0fxols_check_auto_call
Jan 25 11:49:45.291: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_proc
Jan 25 11:49:45.291: htsp_timer - 120000 msechtsp_alert_notify
Jan 25 11:49:45.379: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alerthtsp_call_service_msghtsp_call_service_msg not EFXS (2)
Jan 25 11:49:45.495: htsp_call_bridged invokedhtsp_call_service_msghtsp_call_service_msg not EFXS (2)
Jan 25 11:49:45.503: htsp_process_event: [0/3/0, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
Jan 25 11:49:45.503: htsp_timer_stop
Jan 25 11:49:45.507: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
Jan 25 11:49:45.559: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
Jan 25 11:49:45.559: htsp_timer_stop2
sh voice po su
                                           IN       OUT
PORT            CH   SIG-TYPE   ADMIN OPER STATUS   STATUS   EC
=============== == ============ ===== ==== ======== ======== ==
0/3/0           -- fxo-ls      up    up   idle     off-hook y
0/3/1           -- fxo-ls      up    dorm idle     on-hook y
50/0/1          1      efxs     up    up   on-hook idle     y
50/0/1          2      efxs     up    up   on-hook idle     y
Jan 25 11:50:30.387: htsp_timer_stop3
Jan 25 11:50:30.399: htsp_process_event: [0/3/0, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
Jan 25 11:50:30.399: htsp_timer_stop
Jan 25 11:50:30.399: htsp_timer_stop2
Jan 25 11:50:30.399: htsp_timer_stop3
Jan 25 11:50:30.399: [0/3/0] set signal state = 0x4 timestamp = 0
Jan 25 11:50:30.399: htsp_timer - 300 msec
Jan 25 11:50:30.671: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
Jan 25 11:50:30.699: htsp_process_event: [0/3/0, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
Jan 25 11:50:30.699: fxols_dsp_prealloc_clid_wait. Line reversal alerting DSP preallocation done
Jan 25 11:50:30.699: [0/3/0] htsp_start_caller_id_rx:ETSI_DTMF
Jan 25 11:50:30.699: htsp_start_caller_id_rx create dsp_stream_manager
Jan 25 11:50:30.699: htsp_process_event: [0/3/0, FXOLS_ONHOOK, E_DSP_SIG_0100]
Jan 25 11:50:30.699: [0/3/0] htsp_dsm_create_success returns 1
So what can we do? what is the wronge of my configuration?

anyone can help?

Disconnect problems on FXO Groundstart lines

Hello all,
I have a CCME customer, IOS 12.3-11T2. with 6 FXO ground-start lines on a 2821 platform. They are using PVDM2-32s and two 4Port FXO cards. The call flow goes as follows.
Incoming call from PSTN, hits fxo port which has a connection plar to a number which calls a script in CUE. From the script if you press 0 it calls the reception.
All the above works fine.
If the reception hangs up the call and the PSTN party does not right away, the PSTN party hears a repeating click. If the PSTN then hangs up and you do a show voice port summary it shows as follows
IN OUT
PORT CH SIG-TYPE ADMIN OPER STATUS STATUS EC
========= == ============ ===== ==== ======== ======== ==
0/1/0 -- fxo-gs up up idle on-hook y
Note that the OPER stat shows up and the out status shows on-hook
Until that OPER status changes to DORM every call after the first one will get answered by this port and the PSTN party will hear the repeating click in their ear. It takes up to 30-40 seconds for the call to clear.
If the PSTN party hangs up first the OPER stat will go to DORM almost immediately.
Any thoughts?
Ryan

use the following commands to shorten the time the router waits before determining that the call needs to be dropped due to disconnect tones.
on the voice port
voice-port 1/1/0
timeouts call-disconnect 5
timeouts wait-release 5
these will shorten the IOS default times of 30 sec & 60 sec down to 5 seconds.
use the following link for more info on the FXO disconnect problem.
http://www.cisco.com/en/US/customer/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml

FXO to FXS Back-to-Back Lab

Hi everyone ,
i have the below config in my home lab and its working just fine ... my confusion comes from that FXO is a two stage calling and i have to configure the plar opx to overcome this issue .... but this lab prove the opposite .. i'm able to do successfully calls from both sides without configuring the plar.
i appreciate any one to shed some light on this ..
R1 Config
dial-peer voice 1001 pots
destination pattern 1001
port 3/0/0
dial-peer voice 1 pots
destination-pattern 2...
no digit-strip
port 3/0/1
R2 Config
dial-peer voice 2001 pots
destination-pattern 2001
port 2/0/0
dial-peer voice 1 pot
destination-pattern 1...
no digit-strip
port 2/1/0
thanks a lot.

You are very likely running into an FXO disconnect problem. Take a look at the following URL:
http://www.cisco.com/en/US/tech/tk652/tk653/technologies_tech_note09186a00800ae2d1.shtml
Other URLs that might be helpful are:
http://www.cisco.com/en/US/products/sw/iosswrel/ps5013/products_feature_guide09186a0080080e48.html
http://www.cisco.com/en/US/products/sw/iosswrel/ps1834/products_feature_guide09186a00801601ac.html

FXO not disconnecting after AA picksup

hi all,
I have a CCME10 with ISR2911.
Incoming calls to AA keeps ringing even when caller hangs up the call. I have no issue if it is a direct incoming call to extension(connection plar opx).
Is there any way i can get the call to disconnect correctly with AA?
I tired using "supervisory disconnect anytone" on the voice-port 0/0/0 solved my problem, BUT it messed up my outgoing call. Outgoing call only rang once, then it gets cut off
voice-port 0/0/0
supervisory disconnect dualtone mid-call
supervisory custom-cptone DST
caller-id enable
application
service aa flash:app-b-acd-aa-3.0.0.2.tcl
paramspace english index 1
param number-of-hunt-grps 2
param handoff-string aa
param dial-by-extension-option 1
paramspace english language en
param aa-pilot 6000
paramspace english location flash:
param second-greeting-time 60
param welcome-prompt _bacd_welcome.au
param call-retry-timer 15
param voice-mail 6000
param max-time-call-retry 700
param service-name queue
service queue flash:app-b-acd-3.0.0.2.tcl
param queue-len 10
param aa-hunt3 3333
param number-of-hunt-grps 1
param aa-hunt2 2222
param queue-manager-debugs 1
dial-peer voice 3000 pots
service aa
incoming called-number 6000
port 0/0/0
forward-digits all

This looks like FXO disconnect problem to me:
http://ciscoflair.blogspot.com/2009/05/cisco-fxo-disconnect-issue.html
If you successfully find your disconnect tones for your country I think that your AA is going to work fine also...
HTH,
Dragan

Setting up PLAR with Cisco Unity Connection Call Handler

This is a lab setup and Im doing it to learn. No customer involvement.
Setup
Analog phones - FXS Port - 2951 Router - FXS port - FXO Port - 2951 Router - CUCM 9 - UC 9
|------------PSTN Emulator-------------| |-----MGCP GW----------|
I have a CTI Route Point configured as DN 7000 and it has the default VM profile. The CTI RP is set to FWD All to VM.
The FXO port is set to PLAR to 7000.
When I dial from the PSTN analog phones through the FXO port, I hear the first ring, the FXO port answers, then I hear what sounds like the recorded message beginning to play. Immediately after, I hear the recorded message "You cannot be transferred to this number. Check the number and try again."
I dont do anything and within a second or two, I hear the recorded message for the system call handler start.
I did some more testing. I added an E1 trunk between the PSTN and the MGCP gateway. In the CUCM, I created another CTI Route Point with a DN that I could dial from my PSTN cloud. I also set that CTI Route point up to Call FWD all to VM.
When I dial using the E1 trunk, the call hits the system call handler as expected and I hear my recorded greeting (as expected).
However, calling through the FXS-FXO tie line consistently gets me the error message recording followed by my recorded greeting.
Im currently using loopstart on the tie line, though I have also tried ground start with no difference.
Any ideas?
Jeff

If I understand correctly, you want outside calls to go directly to the call handler but internal calls to ring whatever phone this extension is on.
If I am understanding correctly, then this will probably resolve it. For the purposes of the explanation I will assume that extension 1000 is the number in question:
Create a new partition, we'll just call it ToVM or something like that
Create a new CTI route point with extension 1000 and put it in the ToVM partition, forward all calls for this CTI route point to voicemail
If you don't have one already, create a calling search space for the voice gateway. Call it Gateway-CSS. This should have the same partitions that the gateway can normally call, but it should also have the ToVM partition and that partition should be HIGHER in the list than the partition that has the normal extension 1000 on it.
Apply Gateway-CSS to the gateway
Configure the normal extension 1000 (not in the ToVM partition) the way that you want it to work.
Now when external callers dial 1000 they will go to the call handler because that partition is higher in the CSS and the CTI Route point should be hit first. Internal callers will ring the phone (or whatever it is) because they only have access to the regular internal partition (and not ToVM) that 1000 is in.
This is all assuming that I understand you correctly!

Need to test a bunch of FXO cards.

I am tasked with testing a bunch of FXO cards (VIC-2FXO) and I have a 1760 router with an FXS card in it. I configured the router so that when I call from an IP phone (ext 4000) to the FXS card port 1/0 (ext 4015) it calls the FXO card on port 0/0 which uses plar to ext 4444 which is port 1/1 on the FXS card.
I also tried to plar calls to ext 4001 an IP phone.
When I call 4015 from the IP phone, it rings the 4444 extension once then the FXS port 1/0 (ext 4015) hangs up. A call remains between the FXO port 0/0 and the FXS port 1/1 but since the FXS port 1/0 dropped off you can not talk to the IP phone.
4001 calls ---> 4015 FXS 1/0 which rings ---> FXO 0/0 using PLAR rings ---> 4444 which is on FXS port 1/0
The port light for FXS 1/0 comes on during the call proces but then goes out. The port lights for the FXO 0/0 and FXS 1/1 stay light until I unplug th RJ11 cable between them.
Any ideas how I can keep FSX 1/0 from dropping one the call is answered on FXS 1/1?
Current configuration : 1917 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 15
voice-card 0
voice-card 1
interface FastEthernet0/0
ip address 192.168.69.200 255.255.255.0
speed auto
voice-port 0/0
connection plar 4444
voice-port 0/1
voice-port 1/0
station-id number 4015
voice-port 1/1
station-id number 4444
dial-peer voice 400 pots
destination-pattern 4015
port 1/0
dial-peer voice 200 pots
incoming called-number .
port 0/0
dial-peer voice 444 pots
destination-pattern 4444
port 1/1
telephony-service
max-ephones 6
max-dn 6
ip source-address 192.168.69.200 port 2000
auto assign 1 to 6
max-conferences 4 gain -6
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1 dual-line
number 4000
ephone-dn 2 dual-lin
number 4001
ephone 1
no multicast-moh
mac-address 000A.8AF0.3016
type 7960
button 1:1
ephone 2
no multicast-moh
mac-address 001B.535C.CABD
type 7940
button 1:2
line con 0
line aux 0
line vty 0 4
login
end
Thanks

Michael,
I have seen issues like this and its all dow to actual physical wiring.
Your 2 port FXS & FXO cards have 2 lines available on each port.
Can you ensure that you wire everything as a single 2 wire.
FXS 1/0 RJ11 PIN3 ------------------------------------------------- FXO 0/0 PIN 3
FXS 1/0 RJ11 PIN4 ------------------------------------------------- FXO 0/0 PIN 4
FXS 1/1 RJ11 PIN3 ------------------------------------------------- EXT 4444
FXS 1/1 RJ11 PIN4 ------------------------------------------------- EXT 4444
Do not put through any other wires
Regards,
Alex.
Please rate useful posts.

Urgent Please help with a Conferencing problem in CUCM 6.1

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We have a configuration of CUCM 6.1. We have 24 voice ports (FXO) configured in H 323 registered with the call manger. A CUE 3.2 configured for IVR and mail box. Gateway is a Cisco 3845 and the IOS version is 12.4(15) T10
The problem is whenever a conference is configured with an outside number the FXO port is not releasing after the call disconnects. It is not FXO disconnect problem. This happens only when a conference is taking place. There is no problem with any other outside or inside calls.
I am attaching the configuration of Gateway. Please help me with the problem, I am very much thankful for you that.

The Family pack covers upto 5 computers, other Install Disc cover one computer only.

Direct FXS/E&M integration with external paging amp TOA a-2060 possible?

Hi!
I've never done such integration before so I'd be grateful for any helpful information. So we have a Cisco 2911 ISR G2 router with possible FXS, FXO, E&M cards and an amplifier TOA a-2060 (datasheet attached). The amp has the following input interfaces:
MIC 1, 2: -60dB* (10mV), 600Ω, electronically balanced, screw terminal
TEL: -10dB* (300mV), 10kΩ, transformer isolated balanced, screw terminal
AUX 1, 2: -20dB* (100mV), 10kΩ, unbalanced, RCA pin jack
Is it possible to make this pair work together? And if it is not possible what else do I need to buy in order to make paging work?
Thanks a lot for any help

Thanks for the reply
The "sh ver" definitely shows up the two E&M ports, it's just in CallManager 4.1(3) that I cannot add the VIC2-2E/M module as a subunit to the 2801 gateway.
I'd really like to make this work but will concentrate my efforts on the free FXO port I currently have as it's being recognized (and therefore registered) by CallMananger.

FXS FXO plar problem

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