Tcl & VXML disconnecting
Hello,
Here is my problem, Im using an hybrid application TCL/VXML, with some events I need to call VXML application to send messages to a HTTP server, the problem is that I need to send a message when the call is disconnected, the application doesnt have time to execute the VXML application, Is there a way to stop disconnecting the session after the application is finished ???
This is the debug:
Dec 30 12:02:07.236: //36547//TCL2:/tcl_legObjCmd: leg vxmldialog leg_incoming -u flash://datos_3.vxml -p mensaje3
Dec 30 12:02:07.236: //36547//TCL2:/tcl_vxmlDialogObjCmd: vxmldialog leg_incoming -u flash://datos_3.vxml -p mensaje3
Dec 30 12:02:07.236: //36547//TCL2:/vtd_lg_incoming: argc 6
Dec 30 12:02:07.236: //36547//TCL2:/vtd_lg_incoming: Legs [36547 ]
Dec 30 12:02:07.240: //36547//TCL2:/tcl_parseCallID_vartagObj: VARTAG Translation Leg Count=1
Dec 30 12:02:07.240: //-1//TCL2:/createAVArrayFromTclArray: DuLlama = 2
Dec 30 12:02:07.240: //-1//TCL2:/createAVArrayFromTclArray: DuEsta = 0
Dec 30 12:02:07.240: //-1//TCL2:/createAVArrayFromTclArray: IdLla = 0
Dec 30 12:02:07.240: //-1//TCL2:/createAVArrayFromTclArray: IdTarea = 365471202
Dec 30 12:02:07.240: //-1//TCL2:/createAVArrayFromTclArray: Mensaje = 3
Dec 30 12:02:07.240: //36547//TCL2:/TclInterpDriver: Tcl_Eval Failed in action=act_Cleanup code=1
code=ERROR
Dec 30 12:02:07.240: IVR TCL script failure
Result:
Command Error: vxmldialog incoming leg 36547 has command active
Dec 30 12:02:07.240: IVR TCL script failure errorInfo:
Command Error: vxmldialog incoming leg 36547 has command active
while executing
"leg vxmldialog leg_incoming -u flash://datos_3.vxml -p mensaje3"
(procedure "act_Cleanup" line 20)
invoked from within
"act_Cleanup"
Thx in advance for your help
Its me again,
Now I know that the problem is when I receive "ev_disconnect" the incoming leg just dissapair, so cant execute commands using that leg, is there a way to create a "virtual leg" to use for this ??? I only need VXML to send messages to an http server, no need to play audio or dtmf.
Thx again
Note:
I tried sendmsg but have the same problem, cant use vxml applications without a leg, and dont know how to create ine.
Similar Messages
-
Hi - When VXML G/W sending PING msg to CVP IVR service it using RECOVERY.VXML, Where as i checked in Recovery.vxml it is given as when HTTP not able to communicate with IVR just Handoff the call to Handoff.TCL and it will disconnect the call with cause code 38.
Where as when VXML sending NEW call msg with the help of submitback.template, When i checked in submit back it is given as just Handoff the call to Recovery.VXML, Where as in Recovery it's directing to Handoff.TCL file.
Why it's pointing of using two different files for PING and NEWCALL as a result both ends up with Handoff.TCL
SIVANESAN RHi Senthil Yes, Youu are right, But when you closely look at bootstrap.vxml all it returns when it sengs PING it will try max of 4 times with fetchout value of 7seconds if it doesn't get response as submitback.temp from call server it will handoff to hadoff.TCL where it will disconnect the call with cause code of 38.
But in the case of NEWCALL msg it uses submitback.temp file where it has condition saying that incase of batchfetch just handoff to recovery.vxml, Again Recovery.vxml will handoff the call to Handoff.TCL finally it disconect the call with cause code of 38.
Here my quest is why NEW CALL using recovery.vxml, Instaead it can use Handoff.TCL file right? Is there any difference in that?
SIVANESAN R -
CVP 7.0.2 Abnormally disconnected with error code 38
Hi all,
We have an issue where when called from outbound, it is not possible to reach a certain number. It doesn't even play the dial tone. isdn debug is as follows :
Feb 15 17:51:01.830: ISDN Se0/2/0:15 Q931: TX -> RELEASE pd = 8 callref = 0xE97B
Feb 15 17:51:01.846: ISDN Se0/2/0:15 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x697B
Feb 15 17:51:05.050: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x697D
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98385
Exclusive, Channel 5
Progress Ind i = 0x8083 - Origination address is non-ISDN
Progress Ind i = 0x8A81 - Call not end-to-end ISDN, may have in-band info
Calling Party Number i = 0x0081, '02124783464'
Plan:Unknown, Type:Unknown
Called Party Number i = 0xC1, '2751920'
Plan:ISDN, Type:Subscriber(local)
Feb 15 17:51:05.062: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0xE97D
Channel ID i = 0xA98385
Exclusive, Channel 5
Feb 15 17:51:05.066: %HTTPC-6-REQUEST_FAILED: request URI http://ipphone.fw.garanti.com.tr/ipphone/musteribilgiler?source=TLC&call_data=002124783464-800321200 failed
Feb 15 17:51:05.066: %IVR-3-APP_ERR: Musteri bilgileri serverdan alinamadi - Hata Kodu: error
Feb 15 17:51:05.210: %HTTPC-6-REQUEST_FAILED: request URI http://10.231.51.141:8000/cvp/VBServlet?MSG_TYPE=PING&CALL_DNIS=2403519900&CALL_ANI=sip:[email protected]:5060&ERROR_CODE=0&RECOVERY_VXML=flash:recovery.vxml&CLIENT_TYPE=IOS&CALL_ID=DE8929A61000013C1771025A0AE7338D&SIP_CALL_ID=[email protected]&CALL_UUI=&VERSION=7.0.2 failed
Feb 15 17:51:05.226: %HTTPC-6-REQUEST_FAILED: request URI http://10.231.51.141:8000/cvp/VBServlet?MSG_TYPE=PING&CALL_DNIS=2403519900&CALL_ANI=sip:[email protected]:5060&ERROR_CODE=0&RECOVERY_VXML=flash:recovery.vxml&CLIENT_TYPE=IOS&CALL_ID=DE8929A61000013C1771025A0AE7338D&SIP_CALL_ID=[email protected]&CALL_UUI=&VERSION=7.0.2 failed
Feb 15 17:51:05.230: %HTTPC-6-REQUEST_FAILED: request URI http://10.231.51.141:8000/cvp/VBServlet?MSG_TYPE=PING&CALL_DNIS=2403519900&CALL_ANI=sip:[email protected]:5060&ERROR_CODE=0&RECOVERY_VXML=flash:recovery.vxml&CLIENT_TYPE=IOS&CALL_ID=DE8929A61000013C1771025A0AE7338D&SIP_CALL_ID=[email protected]&CALL_UUI=&VERSION=7.0.2 failed
Feb 15 17:51:05.238: %HTTPC-6-REQUEST_FAILED: request URI http://10.231.51.141:8000/cvp/VBServlet?MSG_TYPE=PING&CALL_DNIS=2403519900&CALL_ANI=sip:[email protected]:5060&ERROR_CODE=0&RECOVERY_VXML=flash:recovery.vxml&CLIENT_TYPE=IOS&CALL_ID=DE8929A61000013C1771025A0AE7338D&SIP_CALL_ID=[email protected]&CALL_UUI=&VERSION=7.0.2 failed
Feb 15 17:51:05.246: %IVR-3-APP_ERR:
**** CVP HANDOFF.TCL: 57080581.76BE11E2.A32B001B.D5293620 abnormally disconnected with code 38. ****
Feb 15 17:51:05.258: ISDN Se0/2/0:15 Q931: TX -> CONNECT pd = 8 callref = 0xE97D
Feb 15 17:51:05.290: ISDN Se0/2/0:15 Q931: RX <- CONNECT_ACK pd = 8 callref = 0x697D
Feb 15 17:51:05.294: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 02124783464 N/A
Feb 15 17:51:07.274: %IVR-3-APP_ERR: ani: 002124783464 dnis: 800321200 | ev_setup_done ls_000 | ev_disconnected di_016 |
Feb 15 17:51:07.298: %ISDN-6-CONNECT: Interface Serial0/2/0:4 is now connected to 02124783464 N/A
Feb 15 17:51:07.298: %ISDN-6-DISCONNECT: Interface Serial0/2/0:4 disconnected from 02124783464 , call lasted 2 seconds
Feb 15 17:51:07.302: ISDN Se0/2/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0xE97D
Cause i = 0x8090 - Normal call clearing
Feb 15 17:51:07.362: ISDN Se0/2/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x697D
Feb 15 17:51:07.366: ISDN Se0/2/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0xE97D
Feb 15 17:51:08.702: ISDN Se0/2/0:15 Q931: RX <- SETUP pd = 8 callref = 0x697E
Sending Complete
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98391
Exclusive, Channel 17
Calling Party Number i = 0x0083, '05336805920'
Plan:Unknown, Type:Unknown
Called Party Number i = 0xC1, '2751935'
Plan:ISDN, Type:Subscriber(local)
Feb 15 17:51:08.714: ISDN Se0/2/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0xE97E
Channel ID i = 0xA98391
Exclusive, Channel 17
Feb 15 17:51:08.718: %HTTPC-6-REQUEST_FAILED: request URI http://ipphone.fw.garanti.com.tr/ipphone/musteribilgiler?source=TLC&call_data=005336805920-800321925 failed
Feb 15 17:51:08.718: %IVR-3-APP_ERR: Musteri bilgileri serverdan alinamadi - Hata Kodu: error
Feb 15 17:51:08.798: ISDN Se0/2/0:15 Q931: TX -> ALERTING pd = 8 callref = 0xE97EDear
Code 38 Indicates that the network is not functioning correctly and that the condition is likely to last
for an extended period.
to me the GW is not able to make SIP connection and is likely due to incorrect
IP Address is seen from the GW.I hope to share the configuration.
Thank you
please rate , if this will help -
Call gets disconnected after playing the menu from CVP
Hello,
When I make a call, the call is going to the ICM and then it is coming to the Call server then to the VXML gateway and then it goes to the VXML server and plays the promt.
The problem i am facing is that once the menu audio is played the call gets disconnected. It doesnt let me choose any options. I feel it is not able to identify the my inputs
I have some debug outputs
Mar 1 01:09:03.095: //-1/C113522D805A/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=1000
----- ccCallInfo IE subfields -----
cisco-ani=sip:[email protected]:5060
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=sip:[email protected];transport=udp
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/cc_api_call_setup_ind_common:
Interface=0x67213130, Call Info(
Calling Number=sip:[email protected]:5060,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=sip:[email protected];transport=udp(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Un
R1#known, FinalDestinationFlag=TRUE,
Incoming Dial-peer=811, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=65
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/ccCheckClipClir:
In: Calling Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Mar 1 01:09:03.103: //-1/C113522D805A/CCAPI/ccCheckClipClir:
Out: Calling Number=sip:[email protected]:5060(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed)
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: :cc_get_feature_vsa malloc success
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: cc_get_feature_vsa count is 1
*Mar 1 01:09:03.103: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
*Mar 1 01:09:03.103: :FEATURE_VSA attributes are: feature_name:0,feature_time:1732532552,feature_id:23
*Mar 1 01:09:03.103: //65/C113522D805A/CCAPI/cc_api_call_setup_ind_common:
Set Up Event Sent;
Call Info(Calling Number=(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=(TON=Unknown, NPI=Unknown))
*Mar 1 01:09:03.111: //65/C113522D805A/CCAPI/cc_process_call_setup_ind:
Event=0x67452800
*Mar 1 01:09:03.119: //65/C113522D805A/CCAPI/ccCallSetContext:
Context=0x65097114
*Mar 1 01:09:03.119: //65/C113522D805A/CCAPI/cc_process_call_setup_ind:
>>>>CCAPI handed cid 65 with tag 811 to app "_ManagedAppProcess_bootstrap"
*Mar 1 01:09:03.143: //65/C113522D805A/CCAPI/ccCallProceeding:
Progress Indication=NULL(0)
*Mar 1 01:09:03.147: //65/C113522D805A/CCAPI/ccCallConnect:
Progress Indication=NULL(0), Data Bitmask=0x0
*Mar 1 01:09:03.151: //65/C113522D805A/CCAPI/ccCallConnect:
Call Entry(Connected=TRUE, Responsed=TRUE)
*Mar 1 01:09:03.239: //65/C113522D805A/CCAPI/cc_api_event_indication:
Event=141, Call Id=65
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_event_indication:
Event Is Sent To Conferenced SPI(s) Directly
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_caps_ind:
Destination Interface=0x0, Destination Call Id=-1, Source Call Id=65,
Caps(Codec=0x0, Fax Rate=0x2, Vad=0x2,
Modem=0x0, Codec Bytes=20, Signal Type=2)
*Mar 1 01:09:03.243: //65/C113522D805A/CCAPI/cc_api_caps_ind:
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),
Playout Max=250(ms), Fax Nom=300(ms))
*Mar 1 01:09:04.163: //65/C113522D805A/CCAPI/ccCallHandoff:
Silent=FALSE, Application=0x6773AB4C, Conference Id=0xFFFFFFFF
*Mar 1 01:09:04.295: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=0(ms), Inter Digit Timeout=0(ms)
*Mar 1 01:09:04.295: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=0(ms), Initial Digit Timeout=0(ms))
*Mar 1 01:09:04.295: //65/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x41, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:04.299: //65/C113522D805A/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=65
*Mar 1 01:09:04.303: //65/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x67213130, callID=0x41, disp=-4, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:04.307: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0xFFFFFFFC, Interface=0x67213130, Call Id=65
*Mar 1 01:09:04.311: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=0(ms), Inter Digit Timeout=0(ms))
*Mar 1 01:09:05.291: //65/C113522D805A/CCAPI/ccAssociateStream:
Coder=5, DTMF Relay=4, Vad=0,
Record Function=0x0, Event Queue=0x66954714, Stream Context=678B0364,
Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:05.295: //65/C113522D805A/CCAPI/ccAssociateStream:
Call Entry(Stream Status=2, Digit Enable=TRUE)
*Mar 1 01:09:05.327: //65/C113522D805A/CCAPI/cc_api_call_associated:
Interface=0x0, CallId=65, Disposition=0
Play Function=0x62C2DFA8, Codec=0x5, Vad=0x0
Media Type=0x3, SPI Context=0x678D2480, Stream CallId=66
TX Dynamic Pt=0x0, RX Dynamic Pt=0x0
*Mar 1 01:09:05.531: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
Initial Digit Timeout=5000(ms), Inter Digit Timeout=10000(ms)
*Mar 1 01:09:05.535: //65/C113522D805A/CCAPI/ccSetDigitTimeouts:
Call Entry(Inter Digit Timeout=10000(ms), Initial Digit Timeout=5000(ms))
*Mar 1 01:09:05.535: //65/xxxxxxxxxxxx/CCAPI/ccCallReportDigits:
(callID=0x41, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:05.535: //65/C113522D805A/CCAPI/ccCallReportDigits:
Enabled=TRUE, Call Id=65
*Mar 1 01:09:05.535: //65/xxxxxxxxxxxx/CCAPI/cc_api_call_report_digits_done:
(vdbPtr=0x67213130, callID=0x41, disp=-4, digit_event=0x1, enable=TRUE, consume=FALSE)
*Mar 1 01:09:05.539: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
Enabled=TRUE, Disposition=0xFFFFFFFC, Interface=0x67213130, Call Id=65
*Mar 1 01:09:05.543: //65/C113522D805A/CCAPI/cc_api_call_report_digits_done:
Call Entry(Initial Digit Timeout=5000(ms), Inter Digit Timeout=10000(ms))
*Mar 1 01:09:08.723: //65/C113522D805A/CCAPI/ccDisassociateStream:
Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:08.723: //65/C113522D805A/CCAPI/ccDisassociateStream:
Call Entry(Stream Status=1, Digit Enable=TRUE)
*Mar 1 01:09:08.735: //65/C113522D805A/CCAPI/ccAssociateStream:
Coder=5, DTMF Relay=4, Vad=0,
Record Function=0x0, Event Queue=0x66954714, Stream Context=678B0364,
Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:08.739: //65/C113522D805A/CCAPI/ccAssociateStream:
Call Entry(Stream Status=2, Digit Enable=TRUE)
*Mar 1 01:09:08.747: //65/C113522D805A/CCAPI/cc_api_call_associated:
Interface=0x0, CallId=65, Disposition=0
Play Function=0x62C2DFA8, Codec=0x5, Vad=0x0
Media Type=0x3, SPI Context=0x678D2480, Stream CallId=66
TX Dynamic Pt=0x0, RX Dynamic Pt=0x0
R1#
R1#
*Mar 1 01:09:16.143: //65/C113522D805A/CCAPI/ccDisassociateStream:
Record Context=0x0, Stream Call Id=66, Call Id=65
*Mar 1 01:09:16.143: //65/C113522D805A/CCAPI/ccDisassociateStream:
Call Entry(Stream Status=1, Digit Enable=TRUE)
*Mar 1 01:09:16.175: //65/C113522D805A/CCAPI/ccCallDisconnect:
Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0)
*Mar 1 01:09:16.179: //65/C113522D805A/CCAPI/ccCallDisconnect:
Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)
*Mar 1 01:09:16.227: //65/C113522D805A/CCAPI/cc_api_call_disconnect_done:
R1#Disposition=0, Interface=0x67213130, Tag=0x0, Call Id=65,
Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)
*Mar 1 01:09:16.235: //65/C113522D805A/CCAPI/cc_api_call_disconnect_done:
Call Disconnect Event Sent
*Mar 1 01:09:16.235: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 1 01:09:16.235: :cc_free_feature_vsa freeing 67445940
*Mar 1 01:09:16.235: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
*Mar 1 01:09:16.235: vsacount in free is 0
I have some basic VXML gateway configuration
R1#show run
Building configuration...
Current configuration : 2588 bytes
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname R1
boot-start-marker
boot-end-marker
no aaa new-model
memory-size iomem 5
ip cef
no ip domain lookup
ip domain name lab.local
ip auth-proxy max-nodata-conns 3
ip admission max-nodata-conns 3
multilink bundle-name authenticated
voice service voip
sip
header-passing
application
service cvperror flash:cvperror.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service new-call flash:bootstrap.vxml
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service ringtone flash:ringtone.tcl
paramspace english language en
paramspace english index 0
paramspace english location flash
paramspace english prefix en
service ringback flash:ringtone.tcl
service handoff flash:handoff.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
service bootstrap flash:bootstrap.tcl
paramspace english index 0
paramspace english language en
paramspace english location flash
paramspace english prefix en
archive
log config
hidekeys
interface FastEthernet0/0
ip address 192.168.1.253 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
no ip http server
no ip http secure-server
control-plane
dial-peer voice 9191 voip
service ringtone
session protocol sipv2
incoming called-number 91919191
codec g711ulaw
no vad
dial-peer voice 9292 voip
service cvperror
session protocol sipv2
incoming called-number 92929292
codec g711ulaw
no vad
dial-peer voice 1503 voip
destination-pattern 1503
session protocol sipv2
session target ipv4:192.168.1.57
dtmf-relay rtp-nte h245-alphanumeric h245-signal
codec g711ulaw
ip qos dscp cs3 signaling
no vad
dial-peer voice 100 voip
destination-pattern 10..
session target ipv4:192.168.1.100
dial-peer voice 811 voip
service bootstrap
session protocol sipv2
session target sip-server
incoming called-number 81T
codec g711ulaw
no vad
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous
line aux 0
exec-timeout 0 0
privilege level 15
logging synchronous
line vty 0 4
login
end
Please help me out how o fix this issue/Thanks for the respose
The microapp what I use is user.microapp.ToExtVXML[]
This is the activity log in the CVP application. I dont see any error logs.
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,newcall,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,ani,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,areacode,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,exchange,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,dnis,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,uui,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.109,,start,iidigits,NA
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.125,CVP Subdialog Start_01,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.593,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.593,Greetings,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.597,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.703,Greetings,exit,done
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.703,Menu,enter,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.707,Menu,interaction,audio_group,initial_audio_group
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,Menu,exit,
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,how,hangup
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,result,normal
192.168.1.57.1368611401109.0.ImationIVR,05/15/2013 15:20:01.875,,end,duration,1
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,newcall,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,ani,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,areacode,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,exchange,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,dnis,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,uui,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,,start,iidigits,NA
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.531,CVP Subdialog Start_01,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.921,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.921,Greetings,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:06.957,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:07.109,Greetings,exit,done
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:20:07.109,Menu,enter,
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,how,app_session_complete
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,result,timeout
192.168.1.57.1368611406531.1.ImationIVR,05/15/2013 15:50:25.718,,end,duration,1819
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,newcall,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,ani,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,areacode,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,exchange,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,dnis,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,uui,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,,start,iidigits,NA
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.312,CVP Subdialog Start_01,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.750,CVP Subdialog Start_01,exit,done
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.750,Greetings,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.782,Greetings,interaction,audio_group,initial_audio_group
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.937,Greetings,exit,done
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:22:56.937,Menu,enter,
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,how,app_session_complete
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,result,timeout
192.168.1.57.1368611576312.2.ImationIVR,05/15/2013 15:53:25.718,,end,duration,1829
Thank you -
CVP 7.0(2) not creating correct string to send to VXML Server
Hello,
When running CVP 7.0(2), has anyone run into an issue with CVP still attaching the [locale]/[app_media_lib] when setting app_media_lib = ".." in the ICM script?
http://1.1.1.1:7000/CVP/en-us/../Server?application=HelloWorld resulted in a bad fetch.
I was able to find a bug that says the workaround is to use "...", but that did not work. Below are the bug details. Any thoughts?
How to configure app_media_lib for VXML Server apps in incorrect
Symptom:
Integrating a Call Studio script with Unified ICME scripts fails
Conditions:
Call returns the following error on the gateway
Loading fail with url (http://{VXML-Server-IP}:7000/CVP/en-us/.../Server?callid={callid}&application={Application Name)
Workaround:
Make sure the app_media_lib ECC Variable is set to "..."
Thanks,
NathanCan you verify from a browser that
http://1.1.1.1:7000/CVP/Server?application=HelloWorld
gets to the first VXML page of the app? does VXMLServer\admin\status.bat show 1 caller in there?
use
http://1.1.1.1:7000/CVP/Server?audium_action=hangup
to disconnect.
Is the ECC variable long enough?
I've never heard of this three dots business.
Regards,
Geoff -
Trying to configure survivability.tcl but no luck
Hi,
I'm trying to configure the survivability.tcl service but I can't seem to get it to work. I've followed the instructions on the CVP config/admin guide but no luck. Attached is the config from a 2811 I'm using on the LAB.
My setup:
ICM 8.5 server running Rogger, Logger, PG, CTIOS, HDS (x.x.x.82)
CVP 8.0 server running CVP call server, vxml server, media server (x.x.x.83)
UCM 8.6 call manager server (x.x.x.81)
2811 running (C2800NM-IPVOICEK9-M) 12.4(24)T5 (x.x.x.254)
4 FXO ports
I would appreciated if someone could look at the config and tell me what's wrong or point me to documentation.
The config is in the attached file.
thanksIt's a bug: CSCth50439 survivability tcl script takes 60 seconds to kick in
I opened a TAC case and the engineer found it right away.
Symptom:
In a Cisco CVP solution installation it might be observed that the survivability tcl script takes 60 seconds to kick in on the gateway when the CVP server is down or unreachable.
Conditions:
When the CVP server is down or unreachable. So far seen in CVP versions 4.1 and 7.0.2.
Workaround:
comment out the following line in survivability.tcl:
# set fsm(any_state,ev_proceeding) "StopTimer same_state"
and load the new file and service on the gateway, after that you can set the following under the survivability service to desired time in seconds to control the time it takes to start the script.
param setup-timeout
thanks again for all the help. -
VXML RINGTONE SERVICE Problem - Call is not routed to agents
We got below mentioned error in CVP Call Server logs for our calls. Due to this calls are not routed to agents.
Apart from our call flow, I need to know what is the cause and solution of this error.
Aborting XFER and disconnecting the caller code 488. RINGTONE SERVICE is not answering within 5000 millisecs, or the caller did not receive or accept the reinvite for ringtone media setup.
1646: 172.20.242.103: May 14 2014 11:50:07.701 +0300: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-69-ICM-561} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C, DLGID = 112 [IVR_LEG] - Processing ,, [MsgBus:CALL_STATE_EVENT], ssId=SYS_IVR1, eventId=DISCONNECT, causeCode=NORMAL_COMPLETION,, LEGID = , DNIS = 9555210577, ANI = sip:[email protected]:5060
1647: 172.20.242.103: May 14 2014 11:50:07.701 +0300: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-69-ICM-561} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C, DLGID = 112 [IVR_LEG] - Publishing ,, [ICM_EVENT_REPORT], dialogueId=112, sendSeqNo=2, eventId=DISCONNECT, causeCode=NORMAL_COMPLETION,, LEGID = , DNIS = 9555210577, ANI = sip:[email protected]:5060
1648: 172.20.242.103: May 14 2014 11:50:07.701 +0300: %CVP_9_0_ICM-7-CALL: {Thrd=pool-1-thread-69-ICM-561} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C, DLGID = 112 [IVR_LEG] - Deleted dialogue. Duration: 0 hrs, 0 mins, 0 secs, 109 msecs
2017: 172.20.242.103: May 14 2014 11:50:07.717 +0300: %CVP_9_0_SIP-7-CALL: {Thrd=DIALOG_CALLBACK.7} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Reinvitation proceeding TRYING.
2018: 172.20.242.103: May 14 2014 11:50:12.685 +0300: %CVP_9_0_SIP-7-CALL: {Thrd=pool-1-thread-66-SIP-7351} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Called ring leg: CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D1273EDA6511E3999FDDDE4246E36C-140005740768575 - [RING-OUT]: status code = 0: elapsed msecs = 5000
2019: 172.20.242.103: May 14 2014 11:50:12.685 +0300: %CVP_9_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Aborting XFER and disconnecting the caller code 488. RINGTONE SERVICE is not answering within 5000 millisecs, or the caller did not receive or accept the reinvite for ringtone media setup. (current=1 max=226) [id:5004]
2020: 172.20.242.103: May 14 2014 11:50:12.685 +0300: %CVP_9_0_SIP-7-CALL: {Thrd=pool-1-thread-74-SIP-7355} CALLGUID = 28D1273EDA6511E3999FDDDE4246E36C LEGID = 28D25F8E-DA6511E3-99A5DDDE-4246E36C - [INBOUND]: Waiting 2000 millisecs before terminating.
69088: 172.20.242.103: May 14 2014 11:50:12.873 +0300: %CVP_9_0_RPT-7-handleFakeNewCall: {Thrd=Thread-58} create fake New Call for >>HEADERS: (JMSType)=MsgBus:VXML_SCRIPT_DETAIL (JMSDestination)=Topic(CVP.VXMLSERVER.REPORT) (JMSTimestamp)=1400057412857 (ServerID)=cvp9lab2.SYS_VXML1:VXML:VXML1:cvp9lab2.MsgBus001 >>BODY: elementName=start elementid=1001871400057412857 timezone=Asia/Riyadh callguid=2BEF7F5ADA6511E399B0DDDE4246E36C localOffset=180 sessionname=172.20.242.103.1400057412842.21144.outbound ani=sip:172.20.243.187 howEventExited=1 sessionvars= sessionid=1001881400057412857 eventExitState=next uui=NA appName=outbound callStartDatetime=Wed May 14 11:50:12 AST 2014 elementTypeID=0 isNewCall=true vxmldatetime=Wed May 14 11:50:12 AST 2014 version=CVP_9_0 calltypeid=6 category=0 iidigits=NA dnis=sip:[email protected]:5060 >>STATE: isTabular=false isWriteable=false cursor=-1
69089: 172.20.242.103: May 14 2014 11:50:12.873 +0300: %CVP_9_0_RPT-7-handleFakeNewCall : {Thrd=Thread-58} 2BEF7F5ADA6511E399B0DDDE4246E36C onHold start time: Wed May 14 11:50:12 AST 2014Hi,
//After that the agent goes to Reserved state but the call doesn't come through.//
I don't have experience on PCCE , but UCCE perspective this looks to me more like ATR(Agent Targeting rule) or Device Target Issue. Ringtone service should not cause Calls to fail.
Can you please post log from CVP, for Particular call that faced this issue?
Regards
Chintan -
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 allThis 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 -
Call drop when VXML server goes down
Can somebody tell me what should ideally happen in the below given scenario?
We have two media/vxml servers. We have a ACE load balancer which sends calls to these two servers. When the customer call is sent to CVP for self service (VXML application) , if the VXML server is shut down (via the CVP OAMP) , should the call drop ? or survivability kicks in the gateway ?
What we saw is the call is dropped after 2-3 seconds of silence.
We see in the CVP PIM logs that " called party disconnect" event and ends the call. Similarly in the gateway we see the transport down message and discnnects the call. is this correct?
Is there any way to preserve this call and treat it?Hi Shravan,
If you had configured survivability then ideally that call should hit the survivabilty. Normally on the VXML Gateways survivability is not auto configured, we would need to configure accordingly depending on the requirement. If survivability is not configured then the call would disconnect
Additionally, could you please let us know the call flow for the self service in your setup
Thanks,
Dass
Please rate useful posts -
Basic tcl and wav files required in VG for a stanalone cvp architecture
hy all
I am very new to this field and still learning can someone help me gain knowledge on this subject my question is
What are the files needed in VG for supporting a standalone CVP architecture ? (in specific the VXML,tcl and wav files ??
What are the use of the following services run in an ICM router ?
rtsvr, dbagent, testsync, router, ccagent, mdsproc
What are the use of the following services run in an ICM logger ?
configlogger, csfs, listlogger, recovery, replicationA stand-alone CVP VXML solution does not need ICM at all. You simply set up the services on the gateway
service CVPSelfService flash:CVPSelfServiceBootstrap.vxml
service helloworld flash:CVPSelfService.tcl
param CVPSelfService-port 7000
param CVPSelfService-app HelloWorld <<<<<<<< whatever the VXML app is called
paramspace english location flash
param CVPPrimaryVXMLServer 164.103.128.20 <<<<<<<< primary VXML server
param CVPBackupVXMLServer 172.20.142.20 <<<<<<<< backup VXML server
I usually also install
service cvperror flash:cvperror.tcl
service ringtone flash:ringtone.tcl
as well.
Regards,
Geoff -
Record time duration for fax in vxml
I am going to get the time duration used for fax sending in vxml:
<transfer name="mycall" destexpr="'fax://' + phone_num" bridge="false" connecttimeout="120s" maxtime="180s" cisco-longpound="true" cisco-mailtoaddress="'xxxx'" >
<catch event="telephone.disconnect.transfer">
<assign name="mydur" expr="mycall$.duration"/>
<goto next="#timeCount"/>
</catch>
</transfer>
<var name="startTime" expr="new Date();"/>
<var name="endTime"/>
<form id="timeCount">
<block>
<assign name="document.startTime" expr="startTime.getTime();"/>
<assign name="document.startTime" expr="Number(startTime);"/>
<assign name="document.endTime" expr="new Date();"/>
<assign name="document.endTime" expr="endTime.getTime();"/>
<assign name="document.endTime" expr="Number(endTime);"/>
<assign name="mydur" expr="document.endTime - document.startTime;"/>
</block>
</form>
But mycall$.duration is 0 after transferring fax.
And even if I use codes to count time duration, the duration, mydur, is about the same regardless of the number of pages sent.
Is it possible to use vxml to record time duration of fax?
Thank you!Paste the following Config:
sh ver
sh run
debug of one call demonstrating the problem :
debug voip ivr all
debug vxml all -
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 -
Hi All,
I have a problem with my vxml inside my TCL script. By the way i'm using the TCL2.0 script. The problem is vxml process change my output to another value. Example :
.Jan 17 13:15:25.185: <var>: namep=aniNo1 expr=0122222222
.Jan 17 13:15:25.185: //-1//VXML:/vxml_expr_eval: expr=var aniNo1=0122222222
.Jan 17 13:15:25.185: <var>: namep=aniNo2 expr=0192505246
.Jan 17 13:15:25.185: //-1//VXML:/vxml_expr_eval: expr=var aniNo2=0192505246
.Jan 17 13:15:25.189: <var>: namep=aniNo3 expr=0361565224
.Jan 17 13:15:25.189: //-1//VXML:/vxml_expr_eval: expr=var aniNo3=0361565224
.Jan 17 13:15:25.189: vxml_counter_reset:
.Jan 17 13:15:25.189: //-1//VXML:/vxml_formitem_select: Status=VXML_STATUS_OK,
.Jan 17 13:15:25.189: //-1//VXML:/vxml_formitem_select: AsyncStatus=VXML_STATUS_OK
.Jan 17 13:15:25.189: //-1//VXML:/vxml_expr_eval: expr=_in76='defined'
.Jan 17 13:15:25.189: //-1//VXML:/vxml_block_proc:
.Jan 17 13:15:25.189: <block>:
.Jan 17 13:15:25.189: //-1//VXML:/vxml_item_attrs_proc: name=_in76
.Jan 17 13:15:25.189: //-1//VXML:/vxml_exit_proc:
.Jan 17 13:15:25.189: <exit>:
.Jan 17 13:15:25.189: //-1//VXML:/vxml_nmtokens_proc: name=aniNo1 name=aniNo2 name=aniNo3
.Jan 17 13:15:25.189: //-1//VXML:/vxml_leave_scope: scope=8
.Jan 17 13:15:25.189: //-1//VXML:/vxml_vapp_terminate: vapp_status=0 ref_count 0
.Jan 17 13:15:25.189: //-1//VXML:/vxml_vapp_terminate: exitstr=aniNo1=21570706,aniNo2=192505246,aniNo3=63367828
.Jan 17 13:15:25.189: //-1//VAPP:/vapp_terminate:
.Jan 17 13:15:25.189: //-1//VAPP:/vapp_session_exit_event_name: Exit Event vxml.session.complete
the number suppose to be 0122222222 but change to 21570706. Can anybody help me? i'm using the vxml code just like in the documentation. Please2 help me? is there any new version of tcl ivr?Hi,
//After that the agent goes to Reserved state but the call doesn't come through.//
I don't have experience on PCCE , but UCCE perspective this looks to me more like ATR(Agent Targeting rule) or Device Target Issue. Ringtone service should not cause Calls to fail.
Can you please post log from CVP, for Particular call that faced this issue?
Regards
Chintan -
%IVR-3-APP_ERR: TCL CVP:
I have a setup where a call comes into our inbound gateway ISDN and then gets routed SIP to CVP and then CVP does a SIP Refer to another number which is handled via a dialpeer on that same gateway. The call completes successfully, but in the inbound gateway I am seeing this error show up for each call:
%IVR-3-APP_ERR: TCL CVP: ******* survivability.tcl: Service 030BB139.243E11D8.84040022.91580338 did not register successfully ******* CallID = 030BB139.243E11D8.84040022.91580338
Any ideas on what is causing this error message?Can anyone help?
I took a look at debugs voice application all. And it is indeed the gateway A survivability tcl disconnecting the call.
TCL CVP: ******* Procedure SetupTimerExpired ******* CallID = D286E3AA.A2A211E4.B5A7A44C.11F01A00
and
TCL CVP: ******* Call setup aborted due to setup timer expiration ******* CallID = D286E3AA.A2A211E4.B5A7A44C.11F01A00
I tried increasing the param setup-timeout 10 as well as param alert-timeout. But same thing happens. I see the call go out tried it to several destinations. Rings once then disconnects. -
I have a query regarding the Cisco IVR Project especially with the TCL Scripts. I use the ?show call application voice summary? (please see figure 1.0).
Questions:
1. Do I need to download the TCL Application in cisco.com and load it to the TFTP Server?
2. Are the TCL Scripts indicated below is built in on the Router? Or it only shows the supported applications that are compatible with the Router?
3. If any case that these TCL are installed on the Router, could it be possible to download it from the router instead of downloading it in cisco.com?
4. Can you help me about TCL Scripts? Though I read some documents there was still some lose ends and loop wholes that I need to clarify with you. Is there any programming (e.g. VXML) in using this? I?m really confused right now.
Figure 1.0
Internap-Philippines# show call application voice summary
name description
CTAPP Call Treatment Action Application
session Basic app to do DID, or supply dialtone.
fax_hop_on Script to talk to a fax redialer
clid_authen Authenticate with (ani, dnis)
clid_authen_collect Authenticate with (ani, dnis), collect if that fails
clid_authen_npw Authenticate with (ani, NULL)
clid_authen_col_npw Authenticate with (ani, NULL), collect if that fails
clid_col_npw_3 Authenticate with (ani, NULL), and 3 tries collecting
clid_col_npw_npw Authenticate with (ani, NULL) and 3 tries without pw
DEFAULT Default system session application
lib_off_app Libretto Offramp
Call Treatment Action Application - Version 1.0.
TCL Script Version 2.0 supported.
TCL Script Version 1.1 supported.
Voice Browser Version 2.0 for VoiceXML 1.0 & 2.0 supported.Hi,
I began working on the B-ACD script and I can now obtain a return status on failure of transfer to an extension.
However, B-ACD uses a different technique for this, rather than setting up a new call leg, and conferencing, it simply hand-offs to the "default" application.
So, changes made to B-ACD may not be good for H.450, and viceversa.
Anyway, once I'm done with the mods, will post the script.
Hope this helps, if so please rate post!
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CUCM installation fails in ESX server
Hi, I'm currently trying to install CUCM 9.1 in ESX server 5.1 Attached snap is the error message and my server specs. Kindly help me on this. Thanks & Regards, Sakthivel M
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ITunes 11.01 playlist view - strange behaviour
iTunes 11.01 playlist view. I've selected a folder and want to create a new playlist. But New / Playlist is greyed out. This often happens. At other times it lets me create, but bounces straight into album view - pretty pictures, but not what I want
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I have no map and can't find how to load it! All I have is a graph with a blue dot for my location...
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Time machine reports that I do not have enough room on my backup disk. There is enough and it will not delete older items. I am using LIon.