CUE AA redirect to external call not disconnecting

Hello,
I have a problem with cue AA redirect to external number not disconnecting only when the external number does not answer; if the external number redirected to does answer, the lines disconnects normally.
My FXO config is as below for all the lines:
voice-port 0/0/0
trunk-group AllFXO
supervisory disconnect dualtone mid-call
input gain 3
output attenuation -1
echo-cancel coverage 32
cptone KW
timeouts initial 5
timeouts interdigit 3
timeouts call-disconnect 3
timeouts ringing 5
timeouts wait-release 1
connection plar 665
impedance complex2
caller-id enable

Hi There,
I noticed that you are getting ISDN cause code 21, not sure why though but the cause code information is interesting...
Cause No. 21 - call rejected.
This cause indicates that the equipment sending this cause does not wish to accept this call. although it could have accepted the call because the equipment sending this cause is neither busy nor incompatible. This cause may also be generated by the network, indicating that the call was cleared due to a supplementary service constraint. The diagnostic field may contain additional information about the supplementary service and reason for rejection.
What it means:
This is usually a telco issue. The call never reaches the final destination, which can be caused by a bad switch translation, or a misconfiguration on the equipment being called.
Now I don't know if it is actually a Telco issue to be honest, because sometimes bad number presentation can cause this issue as well and the carriers switch rejects the call with Cause Code 21 if no other cause code is suitable.
Check your call-forwarded number, make sure you have the outside line number being presented, although it would seem it is anyway otherwise the carrier wouldn't see a call.
Can you please do a "debug isdn q931" and provide us two maybe three examples please
Cheers,
David.

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interface FastEthernet0/0
no ip address
no ip mroute-cache
duplex full
speed 100
interface FastEthernet0/0.100
description SUB-IF for VOICE VLAN
encapsulation dot1Q 100
ip address 10.1.100.1 255.255.255.0
interface FastEthernet0/0.101
description GW FOR VOICEMAIL MODULE AIM/CUE
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discription AIM/CUE VOICEMAIL MODULE
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ip route 10.1.101.2 255.255.255.255 Service-Engin0/1
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session protocol sipv2
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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
service compress-config
service sequence-numbers
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
certificate self-signed 01 nvram:IOS-Self-Sig#3.cer
dot11 syslog
dot11 ssid cisco-data
vlan 1
authentication open
dot11 ssid cisco-voice
vlan 100
authentication open
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
voice call send-alert
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
voice class cause-code 1
no-circuit
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
translate calling 3265
voice translation-profile OUTGOING_TRANSLATION_PROFILE
translate called 1112
voice translation-profile PROFILE_ALL_FXO
translate calling 4
voice translation-profile nondialable
translate called 1000
voice-card 0
dspfarm
dsp services dspfarm
license udi pid UC540W-FXO-K9 sn FHK143074G6
archive
log config
logging enable
logging size 600
hidekeys
username cisco privilege 15 secret 5 $1$vjNa$OFKLhupqR8al6x2b8Xmcj/
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
description $FW_INSIDE$
ip address 10.1.10.2 255.255.255.252
ip access-group 101 in
ip nat inside
ip virtual-reassembly in
interface FastEthernet0/0
description $FW_OUTSIDE$
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
spanning-tree portfast
interface FastEthernet0/1/6
switchport voice vlan 100
macro description cisco-phone
spanning-tree portfast
interface FastEthernet0/1/7
switchport access vlan 20
spanning-tree portfast
interface FastEthernet0/1/8
switchport access vlan 100
macro description cisco-switch
interface Dot11Radio0/5/0
no ip address
shutdown
ssid cisco-data
ssid cisco-voice
speed basic-1.0 basic-2.0 basic-5.5 6.0 9.0 basic-11.0 12.0 18.0 24.0 36.0 48.0 54.0
station-role root
interface Dot11Radio0/5/0.1
encapsulation dot1Q 1 native
bridge-group 1
bridge-group 1 subscriber-loop-control
bridge-group 1 spanning-disabled
bridge-group 1 block-unknown-source
no bridge-group 1 source-learning
no bridge-group 1 unicast-flooding
interface Dot11Radio0/5/0.100
encapsulation dot1Q 100
bridge-group 100
bridge-group 100 subscriber-loop-control
bridge-group 100 spanning-disabled
bridge-group 100 block-unknown-source
no bridge-group 100 source-learning
no bridge-group 100 unicast-flooding
interface Vlan1
no ip address
bridge-group 1
bridge-group 1 spanning-disabled
interface Vlan20
ip address 10.10.10.1 255.255.255.0
interface Vlan100
no ip address
bridge-group 100
bridge-group 100 spanning-disabled
interface BVI1
description $FW_INSIDE$
no ip address
ip nat inside
ip virtual-reassembly in
shutdown
interface BVI100
description $FW_INSIDE$
ip address 10.1.3.1 255.255.255.0
ip nat inside
ip virtual-reassembly in
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http path flash:/gui
ip dns server
ip nat inside source list 1 interface FastEthernet0/0 overload
ip route 0.0.0.0 0.0.0.0 192.168.101.1
ip route 10.1.10.1 255.255.255.255 Integrated-Service-Engine0/0
logging esm config
access-list 1 remark SDM_ACL Category=2
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
access-list 102 permit ip 192.168.101.0 0.0.0.3 any
access-list 103 remark auto generated by SDM firewall configuration##NO_ACES_8##
access-list 103 remark SDM_ACL Category=1
access-list 103 permit tcp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 permit udp 10.1.10.0 0.0.0.3 any eq 2000
access-list 103 deny   ip 10.1.10.0 0.0.0.3 any
access-list 103 deny   ip 192.168.10.0 0.0.0.255 any
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
access-list 103 permit ip any any
access-list 105 permit ip any any
snmp-server community public RO
tftp-server flash:/phones/521_524/cp524g-8-1-17.bin alias cp524g-8-1-17.bin
tftp-server flash:/phones/5x5/spa5x5-7-1-3c.bin alias spa5x5-7-1-3c.bin
tftp-server flash:/phones/525/spa525g-7-4-8.bin alias spa525g-7-4-8.bin
control-plane
bridge 1 route ip
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
voice-port 0/1/0
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
no sip-register
dial-peer voice 5 pots
description ** MOH Port **
destination-pattern ABC
port 0/4/0
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?

Calls from Etisalat PSTN to FXO to voicemail do not disconnect

I have a tricky issue where outside caller calls in and when the call is forwareded to voicemail because of CFNA, the FXO do not disconnect. I have a setup where a Etisalat Analog lines are directly connected to UC560 FXO ports using RJ11.
When a call comes in over the PSTN to an FXO port on my UC560 and the call is answered by the user and after that user goes on-hook, FXO disconnects or gets released normally . When the user does not answer the call and becuse of CFNA timeout the call is forwarded to users voicemail box , then CUE answers, a voicemail is recorded, but when the calling party hangs up FXO doesnot disconnect instead it stays in OFFHOOK state (HANGS). Because of this no more calls are possible on that FXO line. I have to issue shut and no shut command on the FXO to get it released.
The IOS version as follows
uc500-advipservicesk9-mz.151-2.T4
and CME and CUE version are as follows 8.0.2
The follwing is the configuration on my UC560
voice class dualtone-detect-params 1
freq-max-deviation 25
freq-max-power 0
freq-min-power 13
freq-power-twist 4
cadence-variation 4
voice class custom-cptone UAE-CUSTOM-SIEMENS
dualtone disconnect
frequency 425
cadence 425 325 250 500
voice-port 0/1/0
trunk-group ALL_FXO 61
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM-SIEMENS
supervisory dualtone-detect-params 1
no battery-reversal
input gain 14
cptone AE
timeouts call-disconnect 2
timeouts wait-release 2
timing min-ring 62
connection plar opx 202
description Configured by CCA 4 FXO-0/1/0-Custom-BG
caller-id enable
This the dial peer for viocemail
dial-peer voice 2000 voip
description ** cue voicemail pilot number **
translation-profile outgoing XFER_TO_VM_PROFILE
destination-pattern 399
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

HI David,
Here is the debug vpm signal information that i have taken for two scenarios
the configuration on voice-port is as follows
voice class custom-cptone UAE-CUSTOM
dualtone disconnect
frequency 425
cadence 400 350 225 525
voice-port 0/1/1
translation-profile incoming INCOMING_CallerID_PROFILE
supervisory disconnect dualtone mid-call
supervisory custom-cptone UAE-CUSTOM
input gain 14
cptone AE
timeouts call-disconnect 2
timeouts wait-release 2
connection plar opx 202
description Configured by CCA 4 FXO-0/1/1-Custom-OP
caller-id enable
the came same configuration above with battery reversal answer but no use sitll same issue.
The other tricky thing that is happening is when the call is forwarded to voicemail of the user and after the external caller disconnects the FXO on UC540 does not disconnect immediately, instead it disconnects after the default messgae size is reached. ie the default message size of voicemail box is 240 sec so after 240 sec the FXO port is released or disconnects and a large amount of silence is being recorded in the users mailbox for about 240 seconds.
the following is the debug capture taken
========================================================================================================================
When the call comes in and call is forwarded to voicemail because of CFNA on the user phone
UC_540#
000691: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_DSP_SIG_0000]fxols_onhook_ringing
000692: htsp_timer - 125 msec
000693: htsp_process_event: [0/1/1, FXOLS_WAIT_RING_MIN, E_HTSP_EVENT_TIMER]fxols_wait_ring_min_timer
000694: htsp_timer - 10000 msec
000695: htsp_timer3 - 5600 msec
000696: [0/1/1] htsp_start_caller_id_rx:BELLCORE
000697: htsp_start_caller_id_rx create dsp_stream_manager
000698: [0/1/1] htsp_dsm_create_success returns 1
UC_540#
000699: htsp_process_event: [0/1/1, FXOLS_RINGING, E_DSP_SIG_0100]
000700: fxols_ringing_not
000701: htsp_timer_stop
000702: htsp_timer - 10000 msec
000703: [0/1/1] htsp_dsm_feature_notify_cb returns 2 id=DSM_FEATURE_SM_CALLERID_RX
000704: htsp_process_event: [0/1/1, FXOLS_RINGING, E_HTSP_CALLERID_RX_DONE]
000705: htsp_timer_stop
000706: htsp_timer_stop3
000707: [0/1/1] htsp_stop_caller_id_rx. message length 25htsp_setup_ind
000708: [0/1/1] get_fxo_caller_id:Caller ID received. Message type=128 length=25 checksum=B1
000709: [0/1/1] Caller ID String 80 16 01 08 30 34 31 39 31 33 34 30 02 0A 30 35 30 39 35 37 38 33 30 39 B1
000710: [0/1/1] get_fxo_caller_id calling num=0509856909 calling name= calling time=04/19 13:40
000711: fxols_callerid_done: call being answered
000712: [0/1/1] htsp_dsm_close_done
000713: htsp_process_event: [0/1/1, FXOLS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]
000714: fxols_wait_setup_ack:
000715: htsp_timer - 6000 msec
000716: htsp_timer_stop
UC_540#3
000717: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_HTSP_PROCEEDING]fxols_offhook_prochtsp_setup_req
000718: htsp_process_event: [50/0/30.1, EFXS_ONHOOK, E_HTSP_SETUP_REQ]efxs_onhook_setup
000719: htsp_ephone_start_caller_id_tx calling num=90509578309 calling name = called num=201 orig called num=
000720: [50/0/30.1] set signal state = 0x0 timestamp = 0
000721: efxs_onhook_setup: local target is available
htsp_alerthtsp_alert_notify
000722: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_HTSP_ALERT]fxols_offhook_alert
UC_540#
000723: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000724: htsp_timer_stop
000725: htsp_timer_stop2
UC_540#
000726: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000727: htsp_timer_stop2
000728: htsp_timer - 6000 msec
UC_540#
000729: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000730: htsp_timer_stop
000731: htsp_timer_stop2
UC_540#
000732: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000733: htsp_timer_stop2
000734: htsp_timer - 6000 msec
UC_540#
000735: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000736: htsp_timer_stop
000737: htsp_timer_stop2
UC_540#
000738: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000739: htsp_timer_stop2
000740: htsp_timer - 6000 msec
UC_540#
000741: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0000]fxols_proceed_ring
000742: htsp_timer_stop
000743: htsp_timer_stop2
UC_540#
000744: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_DSP_SIG_0100]fxols_proceed_clear
000745: htsp_timer_stop2
000746: htsp_timer - 6000 msec
UC_540#
000747: htsp_timer_stop3
000748: htsp_process_event: [50/0/30.1, EFXS_WAIT_OFFHOOK, E_HTSP_RELEASE_REQ]efxs_waitoff_release
000749: [50/0/30.1] set signal state = 0x4 timestamp = 0
000750: htsp_call_bridged invoked
000751: htsp_process_event: [0/1/1, FXOLS_PROCEEDING, E_HTSP_CONNECT]fxols_offhook_connect
000752: [0/1/1] set signal state = 0xC timestamp = 0
000753: htsp_timer_stop
000754: htsp_process_event: [0/1/1, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
000755: htsp_process_event: [0/1/1, FXOLS_CONNECT, E_DSP_SIG_0110]fxols_rvs_battery
000756: htsp_timer_stop2
000757: htsp_timer_stop2
UC_540#
After the voicemail box, default message size is reached ie after 240 seconds the FXO port disconnects and following is the continuation of debug vpm signal cmd.
UC_540#
000758: htsp_timer_stop3 htsp_setup_req
000759: htsp_process_event: [50/0/300.1, EFXS_ONHOOK, E_HTSP_SETUP_REQ]efxs_onhook_setup
000760: htsp_ephone_start_caller_id_tx calling num=399 calling name = called num=A800201 orig called num=
000761: [50/0/300.1] set signal state = 0x0 timestamp = 0
000762: efxs_onhook_setup: local target is available
htsp_alerthtsp_call_feature:feature 25
htsp_call_feature: caller id enable 0x3 call_connected 0
000763: htsp_process_event: [50/0/300.1, EFXS_WAIT_OFFHOOK, E_HTSP_CALLERID_WAITING]
000764: efxs_callerid_update
000765: efxs_callerid_update process caller_id_string
000766: efxs_callerid_update process caller_id_string OK
UC_540#
000767: efxs_callerid_update number= [399] name= []
UC_540#
000768: htsp_timer_stop3
000769: htsp_process_event: [0/1/1, FXOLS_CONNECT, E_HTSP_RELEASE_REQ]fxols_offhook_release
000770: htsp_timer_stop
000771: htsp_timer_stop2
000772: htsp_timer_stop3
000773: [0/1/1] set signal state = 0x4 timestamp = 0
000774: htsp_timer - 2000 msec
000775: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_DSP_SIG_0110]
UC_540#
000776: htsp_process_event: [0/1/1, FXOLS_GUARD_OUT, E_HTSP_EVENT_TIMER]fxols_guard_out_timeout
000777: htsp_process_event: [0/1/1, FXOLS_ONHOOK, E_DSP_SIG_0100]
000778: htsp_timer_stop3
000779: htsp_process_event: [50/0/300.1, EFXS_WAIT_OFFHOOK, E_HTSP_RELEASE_REQ]efxs_waitoff_release
000780: [50/0/300.1] set signal state = 0x4 timestamp = 0
UC_540#
Can any one let me know what is happing here. when the call is forwarded to voicemail of the user, why FXO Port on UC540 not getting disconnected soon after the external caller disconnects the call.and insted it disconnects approximately after 240 seconds of call forwarded to voicemail.
I tried the same cofiguration as above with battery reversal answer in voice-port configuration but no use sitll same issue.

Spa232d not disconnecting call

Hi,
We replaced our SPA3102 with a SPA232D.
FXO port is connected to our analog line, Internet port is connected to our LAN.
The PSTN line on the SPA232D is registered with our Asterisk server so calls go through our analog line.
From our IP phones to outside, everything is OK.
From outside to our IP phones, there is some problems when the call is ended : if we hang up before our external caller, the line remains open on the caller's phone. If the caller ends the call before us, we hear the disconnect tone and the calls end normally.
We tried some settings here and there but unsuccesfull... Could you help please?
Thanks!
Thomas

Could we have a bit more information please: what version of iTunes and firmware are you running; 7.5/7.6 and 1.1.3/1.1.3?
Also, is the iPhone locked to a carrier; and when did this problem start? After any updates?
Cheers.

Feature 'nested external environment calls' not implemented

Hi.
I've created a .Net 3.5 DLL which is called by a stored procedure using CLR in a trigger.
Inside one of the functions I've got an update-statement against the same table as the trigger which calls the CLR code.
That update results in a SAExeption:
30.04.2014 14:19:55 (ConnID. 89) - W747lnr=4000996, Error: Feature 'nested external environment calls' not implemented
iAnywhere.Data.SQLAnywhere.SAException: Feature 'nested external environment calls' not implemented
   at iAnywhere.Data.SQLAnywhere.SACommand.ExecuteReader()
   at CLRKalkyle.Exts.Execute(SAConnection conn, String sql, Object[] args)
   at CLRKalkyle.CalculusFunctions.DivInnfrakt(SAConnection conn, Pallet palle, Decimal totkvant)
   at CLRKalkyle.CalculusFunctions.GetInnfrakt(SAConnection conn, Int32 klientnr, String uttakstype, Int32 uttaksnr, Int32 w722lnr)
   at CLRKalkyle.Exts.DivKalkyle(SAConnection conn, Int32 klientnr, String uttakstype, Int32 uttaksnr, Int32 w722lnr, Int32 w747lnr)
   at CLRKalkyle.CLRKalkyle.GenKalkyle(Int32 klientnr, Int32 w747lnr)
I'm using SAServerSideConnection without transaction (the CLR is called per row in the database).
Is it possible to run the update in a new thread, or perhaps a new connection?
Or do I have to run the update after all the other code is finished?
Hope you understand my problem.
Regards,
Bjarne Anker

This is the trigger:
ALTER TRIGGER "wau_w747_maaoppdat_on" after update of MaaOppdateres
order 2 on MTS.W747KalkyleData
referencing old as old_name new as new_name
for each row when(old_name.MaaOppdateres = 0 and new_name.MaaOppdateres = 1)
begin
    call spw_GenKalkyle_w747lnr(new_name.klientnr,new_name.w747lnr)
end
Stored procedure:
ALTER PROCEDURE "MTS"."spw_GenKalkyle_w747lnr"( in p1 integer,in p2 integer )
external name 'D:\\SVN\\Trading\\CLRKalkyle\\CLRKalkyle\\bin\\Release\\CLRKalkyle.dll::CLRKalkyle.CLRKalkyle.GenKalkyle(int,int)' language clr
This is where the exception occurs in the program:
sql = @"update w747kalkyledata w747
set w747.maaoppdateres = 1
from w722fakturalinje w722
where w722.KlientNr = ?
and w722.TilgangsType = ?
and w722.TilgangsNr = ?
and w722.tilgangsnr > 0
and w722.klientnr = w747.klientnr
and w722.uttakstype = w747.uttakstype
and w722.uttaksnr = w747.uttaksnr
and w722.w722lnr = w747.w722lnr
and w722.uttakstype = 'DF'
and w747.maaoppdateres = 0";
SAServerSideConnection.Connection.Execute(sql, palle.klientnr, palle.tilgangstype, palle.tilgangsnr);
In short, the trigger is at the table W747 and the code fails when running the update against the same table.
I guess it would be best to send the sourcecode via email if you need to look at the whole picture.
Regards,
Bjarne Anker

Strom 9520 , not disconnecting calls.

Hi,
I bought first blackberry phone, Strom2 .
It works quite fine but sometimes when I make a call its not disconnecting them. I have to take out battery and then use it for 1 day and then again the same problem
During this, My whole screen get locked and I cant make any click on screen. can you please tell me whats the problem?
[url=www.soundshop.si/kitare]kitare[/url]

shiwak,
Perform a reload or upgrade the BlackBerry Device Software. For more information on this procedure see the following article:
http://www.blackberry.com/btsc/KB03901
-FB
Come follow your BlackBerry Technical Team on Twitter! @BlackBerryHelp
Be sure to click Kudos! for those who have helped you.
Click "Accept as a Solution" for posts that have solved your issue(s)!

First Redirect Number (external) and call localization

Hi
Does First Redirect Number (external) on the MGCP Gateway support some type of call localization?.
Thanks
Alex

If you are running CCM 4.1, I think you should check the "Block OffNet To OffNet Transfer*" under the Service-> Service Parameters-> CallManager to "False"
Thanks

Torch 9860 does not disconnect after ending call

Hi,
I have been having this peculiar problem with my blackberry torch 9860 smartphone for the past 1 week. After finishing a voicecall, in many situations the phone does not disconnect even if I press the end call button repeatedly. It just seems to "hang" for 30-40 seconds and then disconnect. Sometimes it dosconnects only if the person at the other end hangs up.
Kindly help out in solving this issue

Try a Battery pull please, with the BlackBerry device powered ON, remove the battery 15 seconds and then reinsert the battery to reboot device. This will clear all cache like rebooting a PC. Then try again and see if your problem persists.
Click here to Backup the data on your BlackBerry Device! It's important, and FREE!
Click "Accept as Solution" if your problem is solved. To give thanks, click thumbs up
Click to search the Knowledge Base at BTSC and click to Read The Fabulous Manuals
BESAdmin's, please make a signature with your BES environment info.
SIM Free BlackBerry Unlocking FAQ
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BES 12 and BES 5.0.4 with Exchange 2010 and SQL 2012 Hyper V

External Calling

Hi I'm trying to enable external calls currently I can call internally and receive external however I want to be able to phone out on one if not all phones connected to the network.
Below is my current config:
Building configuration...
Current configuration : 5655 bytes
version 15.5
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
bsd-client server url https://cloudsso.cisco.com/as/token.oauth2
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-participate wic 2
network-clock-participate wic 3
ip dhcp excluded-address 192.168.0.1 192.168.0.50
ip dhcp excluded-address 192.168.0.241 192.168.0.255
ip dhcp pool PHONES
network 192.168.0.0 255.255.255.0
default-router 192.168.0.1
dns-server 192.168.0.5 192.168.0.6
option 150 ip 192.168.0.17
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type basic-net3
cts logging verbose
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 192.168.0.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
h323
h245 tunnel disable
sip
bind control source-interface GigabitEthernet0/0
registrar server expires max 3600 min 3600
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729br8
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 4
call start fast
voice hunt-group 1 longest-idle
timeout 0
license udi pid CISCO2901/K9 sn FGL173220Y3
license accept end user agreement
hw-module ism 0
hw-module pvdm 0/0
hw-module pvdm 0/1
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 192.168.0.17 255.255.255.0
duplex auto
speed auto
interface ISM0/0
description Unity-Express-Module
ip unnumbered GigabitEthernet0/0
ip virtual-reassembly in
service-module ip address 192.168.0.10 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 192.168.0.17
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface ISM0/1
no ip address
interface BRI0/0/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/0/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/1/0
description Test_ISDN_1
no ip address
isdn switch-type basic-net3
isdn overlap-receiving
isdn point-to-point-setup
isdn layer1-emulate network
isdn incoming-voice voice
isdn static-tei 0
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 192.168.0.10 255.255.255.255 ISM0/0
control-plane
voice-port 0/0/0
no vad
compand-type a-law
no comfort-noise
cptone GB
description TEST_ISDN
voice-port 0/0/1
input gain -6
output attenuation -6
echo-cancel coverage 64
no vad
compand-type a-law
no comfort-noise
cptone GB
timeouts interdigit 6
description ***2BRI-NT/TE Port***
bearer-cap Speech
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/2/0
voice-port 0/2/1
voice-port 0/3/0
voice-port 0/3/1
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dial-peer voice 2 voip
destination-pattern 1[0-9][0-9][0-9]
session target ipv4:162.168.0.17
dial-peer voice 1 pots
destination-pattern 9.
port 0/0/0
forward-digits all
dial-peer voice 3 pots
destination-pattern 9T
port 0/0/1
gatekeeper
shutdown
telephony-service
max-ephones 20
max-dn 200
ip source-address 192.168.0.17 port 2000
network-locale GB
load 7906 term11.default
load 7960-7940 P0030801SR02
load 6921 SCCP69xx.9-2-1-0
load 6941 SCCP69xx.9-2-1-0
max-conferences 8 gain -6
web admin system name alex secret 5 $1$eSOq$7L1LD5IBwR3F5v0.3gc3i1
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 1001
name ***Black 6941 ***
ephone-dn 2
number 1002
name ***6921 v1***
ephone-dn 3
number 1003
name ***6921 v2***
ephone-dn 4
number 1004
name ***Big Bertha***
ephone 1
mac-address E8B7.484E.8483
type 6941
button 1:1
ephone 2
mac-address 8478.ACC7.13C0
type 6941
button 1:2
ephone 3
mac-address 8478.ACC7.13A4
type 6921
button 1:3
ephone 4
mac-address 0023.5E18.A3AA
type 7940
button 1:4
ephone-hunt 1 longest-idle
pilot 619879
list 1001, 1002, 1003
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login
transport input none
scheduler allocate 20000 1000
end

Have tried adding however still getting an engaged tone. Below is my latest config can anyone help?
Building configuration...
Current configuration : 5860 bytes
! Last configuration change at 10:37:45 UTC Wed Jan 21 2015
version 15.5
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
bsd-client server url https://cloudsso.cisco.com/as/token.oauth2
network-clock-participate wic 0
network-clock-participate wic 1
network-clock-participate wic 2
network-clock-participate wic 3
ip dhcp excluded-address 192.168.0.1 192.168.0.50
ip dhcp excluded-address 192.168.0.241 192.168.0.255
ip dhcp pool PHONES
network 192.168.0.0 255.255.255.0
default-router 192.168.0.1
dns-server 192.168.0.5 192.168.0.6
option 150 ip 192.168.0.17
ip cef
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type basic-net3
cts logging verbose
voice-card 0
dspfarm
dsp services dspfarm
voice call send-alert
voice call carrier capacity active
voice rtp send-recv
voice service voip
ip address trusted list
ipv4 192.168.0.0 255.255.255.0
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
redirect ip2ip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
modem passthrough nse codec g711ulaw
h323
h245 tunnel disable
sip
bind control source-interface GigabitEthernet0/0
registrar server expires max 3600 min 3600
voice class codec 1
codec preference 1 g711alaw
codec preference 2 g729br8
codec preference 3 g729r8
voice class h323 1
h225 timeout tcp establish 4
call start fast
voice hunt-group 1 longest-idle
timeout 0
license udi pid CISCO2901/K9 sn FGL173220Y3
license accept end user agreement
hw-module ism 0
hw-module pvdm 0/0
hw-module pvdm 0/1
redundancy
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address 192.168.0.17 255.255.255.0
duplex auto
speed auto
interface ISM0/0
description Unity-Express-Module
ip unnumbered GigabitEthernet0/0
ip virtual-reassembly in
service-module ip address 192.168.0.10 255.255.255.0
!Application: CUE Running on ISM
service-module ip default-gateway 192.168.0.17
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
interface ISM0/1
no ip address
interface BRI0/0/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/0/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/1/0
description Test_ISDN_1
no ip address
isdn switch-type basic-net3
isdn overlap-receiving
isdn point-to-point-setup
isdn layer1-emulate network
isdn incoming-voice voice
isdn static-tei 0
interface BRI0/1/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/2/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/0
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface BRI0/3/1
no ip address
isdn switch-type basic-net3
isdn point-to-point-setup
interface Vlan1
no ip address
ip forward-protocol nd
no ip http server
no ip http secure-server
ip route 192.168.0.10 255.255.255.255 ISM0/0
control-plane
voice-port 0/0/0
no vad
compand-type a-law
no comfort-noise
cptone GB
description TEST_ISDN
voice-port 0/0/1
input gain -6
output attenuation -6
echo-cancel coverage 64
no vad
compand-type a-law
no comfort-noise
cptone GB
timeouts interdigit 6
description ***2BRI-NT/TE Port***
bearer-cap Speech
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/2/0
voice-port 0/2/1
voice-port 0/3/0
voice-port 0/3/1
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
dial-peer voice 2 voip
destination-pattern 1[0-9][0-9][0-9]
session target ipv4:162.168.0.17
dial-peer voice 1 pots
destination-pattern 9.
port 0/0/0
forward-digits all
dial-peer voice 3 pots
destination-pattern 9T
port 0/0/1
dial-peer voice 100 pots
description outbound dialpeer 1
preference 7
destination-pattern 1[2-9].........
port 0/0/0
forward-digits all
gatekeeper
shutdown
telephony-service
max-ephones 20
max-dn 200
ip source-address 192.168.0.17 port 2000
network-locale GB
load 7906 term11.default
load 7960-7940 P0030801SR02
load 6921 SCCP69xx.9-2-1-0
load 6941 SCCP69xx.9-2-1-0
max-conferences 8 gain -6
web admin system name alex secret 5 $1$eSOq$7L1LD5IBwR3F5v0.3gc3i1
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-dn 1
number 1001
name ***Black 6941 ***
ephone-dn 2
number 1002
name ***6921 v1***
ephone-dn 3
number 1003
name ***6921 v2***
ephone-dn 4
number 1004
name ***Big Bertha***
ephone 1
mac-address E8B7.484E.8483
type 6941
button 1:1
ephone 2
mac-address 8478.ACC7.13C0
type 6941
button 1:2
ephone 3
mac-address 8478.ACC7.13A4
type 6921
button 1:3
ephone 4
mac-address 0023.5E18.A3AA
type 7940
button 1:4
ephone-hunt 1 longest-idle
pilot 619879
list 1001, 1002, 1003
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
login
transport input none
scheduler allocate 20000 1000
end

CUE AA redirect to external call not disconnecting

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