Transfered calls are failing

Hi All,
In IPCCE environment, some of the agent to agent transfer calls are failing with the below error.
Couldn't find CallConnection with DeviceID of 32767308. CILConnectionID:9
I can see this same device ID 32767308 for all the failed calls. This device ID 32767308 is present only in ctios client and server logs. Any idea where/why this device ID getting generated for the failed calls?
Regards,
Adithya

Hi Irfan,
Thanks for the reply.
Let me explain the cal flow.
Agent A selects the available agents(Agent B) first and then consults with agent B(First Call leg). If agent B agrees to take the call, agent A does the complete transfer.(call leg 2) which triggers the ICM script thru a Dailed Number.
Just we found that here the call leg 1 is getting dropped before establishing the call leg 2. Thats why we see that the error message Couldn't find CallConnection with DeviceID. Any suggestion?
Regards,
Adithya

Similar Messages

PCS calls are failing through "SendToVRU" node.

Hi All
we are using icm enabled PCS and getting problem that calls are failing through the send to VRU node.
Actually we are getting Handoff.tcl errors on our monitoring system. From CVP error log we found that those errors are against PCS calls only and from ICM script we found the above.For reference i am attaching screenshot.
Kindly help if anyone faced the same issue and got the solution.

Not much info is provided, but are the calls which failing belong to a particular language? check below mentioned link and see if it applies for you.
https://tools.cisco.com/bugsearch/bug/CSCtk13452/?referring_site=ss

UCCE calls are failing at the RunExternalScript node

Dear members,
It is a UCCE Lab environment with CVP version 9.0
My issue is that all the calls are failing at the RunExternalScript node.
In the CVP logs, the below error appears.
External VXML located at http://192.168.2.215:7000/CVP/en-us/app/Server?_dnis=3000&application=Test&callid=565&_ani=565 resulted in a bad fetch
Attached are the ICM script, CVP script, CVP logs and rtr logs.
Is there any variable that needs to be added in the ICM script?
Thank you in advance for your help.
Lara

Thank you Jameson for replying.
I was using the variable user.microappp.sys_media_lib instead of user.microapp.app_media_lib.
But even after I corrected this variable, I'm still facing a bad fetch error.
70: 192.168.2.215: Mar 09 2015 11:24:49.294 +0200: %CVP_9_0_IVR-3-CALL_ERROR: CALLGUID=0145638000010000000000421500A8C0 DNIS=012345678918 CVP VXML Server encountered a Bad-Fetch Error - URL: http://192.168.2.215:7000/CVP/en-us/../Server?_dnis=3000&application=Test&callid=565&_ani=565 (Client: 10.1.1.103) [id:3023]
71: 192.168.2.215: Mar 09 2015 11:24:49.294 +0200: %CVP_9_0_IVR-3-CALL_ERROR: RunScript Error from 10.1.1.103 [CVP_BAD_FETCH(45)] CALLGUID: 0145638000010000000000421500A8C0 DNIS=012345678918 {VRUScriptName: 'GS,Server,V' ConfigParam: ''} [id:3023]
108: 192.168.2.215: Mar 09 2015 11:24:49.310 +0200: %CVP_9_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 0145638000010000000000421500A8C0 LEGID = 50331337-C57A11E4-BCB6D284-70B23AF8 - [INBOUND] - DIALOGUE_FAILURE from ICM Router sends 404 rejection to call. errorcode=15 [id:5004]
111: 192.168.2.215: Mar 09 2015 11:24:49.310 +0200: %CVP_9_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 0145638000010000000000421500A8C0 LEGID = 50331337-C57A11E4-BCB6D284-70B23AF8 - [INBOUND] - ABNORMALLY ENDING - SIP code [404], Reason Hdr [SIP;cause=404] Not Found, GW call using SURV TCL flag [false], NON NORMAL flag [true], DNIS [3000], ANI [565] with AGE (msecs) 54334 and Call History : 012345678918|-1; [id:5004]
120: 192.168.2.215: Mar 09 2015 11:24:50.152 +0200: %CVP_9_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 0145638000010000000000421500A8C0 LEGID = 50331337-C57A11E4-BCB6D284-70B23AF8 - [INBOUND]: Refer failed with 503 - Service Unavailable. May be a problem with Routing Configuration or Gateway Dial-Peer. [id:5004]
122: 192.168.2.215: Mar 09 2015 11:24:50.152 +0200: %CVP_9_0_SIP-3-SIP_CALL_ERROR: CALLGUID = 0145638000010000000000421500A8C0 LEGID = 50331337-C57A11E4-BCB6D284-70B23AF8 - [INBOUND] - ABNORMALLY ENDING - SIP code [404], Reason Hdr [SIP;cause=404] Not Found, GW call using SURV TCL flag [false], NON NORMAL flag [true], DNIS [3000], ANI [565] with AGE (msecs) 55176 and Call History : 012345678918|-1; [id:5004]
I can play the .wav file from the browser but the calls are still failing at the Run External Script node.
I restarted the CVP Call/VXML server but with no luck.
Your help is highly appreciated.
Thank you.
Lara

My facetime calls are failing from my MAC

Both received and sent calls from my MAC are failing, although prior to today they've always worked. HELP

Morning sandyab60,
Article: TS4185 FaceTime for Mac: Troubleshooting FaceTime should help you get FaceTime working again.
Hope this helps,
Mario

Transferred calls are dropped after CNA timeout

I have a problem I just can't find the answer to. I am certain tha the answer would be easy, but right now, I'm lost.
Here is a quick description of the problem (CCM 4.1.3 btw.)
If a call to DN 1111 in partition X is not answered (internal and external), it is redirected to the operator at DN 3001 in partition Y.
However, if a call to DN 2222 is answered, then manually transfered to DN 1111 and not answered. The call is dropped (and a busy signal is issued) when it is supposed to redirect to DN 3001.
The strange thing is: if I but the DN 3001 in partition X (Same as DN 1111), the redirection works.
Short summary: When a direct inbound call to any DN is timed out and then redirected to another DN. It works.
When a call has been transfered to a DN, and then is timed out. Redirection is only successfull within the "current" partition.
Which CSS is used when a call is transfered, and then redirected (timed out). It is clear to me, that it does not use the CSS defined in line-config of the phone.
Any good answers?
Kind Regards
Lasse

Yes it is. I will try to clearify the issue:
If I dial an internal DN (DN 2003 dials DN 2004 in partition X)), and the dialled party does not pick up the phone. The line is confgured to FWD No Answer Internal (and eksternal) to DN 3001 (in partition Y) using a CSS with partition Y in it. There is nothing wrong with this scenario, and the call is forwarded to the swichboard as designed.
But if DN 2001 calls DN 2002 and the user at DN 2002 manually transfer the call to DN 2003 (and complets the transfer without waiting for DN 2003 to pick up the call), the FWD No Answer Internal (and eksternal) to DN 3001 fails. The same rule is applied, but gets disconnected when beeing transfered to a different partition. If DN 3001 is put in partition X (with DN 200X) the FWD No Answer Internal (and eksternal) to DN 3001 works just fine.

Transfer a Transferred Call failed

I have issue with Lync, transfer a transferred call always failed, the scenario as following:
User A calls User B (Success)
User B transfer to User C (Success)
User C transfer to User D (Failed)
This issue happens always for all the users whether the call is Lync-to-Lync or PSTN-to-Lync call.
Any help is appreciated

Hi,
Did all user in one site or in different sites? Since when calls are transferred or forwarded, the routing of calls is affected by Location-Based Routing.
You can refer to the link below of “Deployment Process for Location-Based Routing”:
http://technet.microsoft.com/en-us/library/jj994055.aspx
Please also try to disable REFER support. From the Control Panel -> Voice Routing -> Trunk Configuration, open your trunk for the gateway (if you don't have one, open the Global one) and change REFER support to None.
Click OK. Click Commit and commit the changes.
Wait a few minutes and try again.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Take back a transferred call

Hi,
Is it possible to take back a call that has been transferred? Take this example:
1. A customer calls employee A
2. Employee A answers the call and decides to transfer the call to employee B
3. Employee B does not answer the call so Employee A wants to take back the call
Is that possible in any way?

Hi,
Please set Lync client policy with the parameter AttendantSafeTransfer, set the value to “True”. When set to True, Attendant operates in "safe transfer" mode; this means that transferred calls that do not reach the intended recipient
will reappear in the incoming area along with a "Failed Transfer" notice. When set to False, transferred calls that fail to reach the intended recipient will not reappear in the incoming area.
Best Regards,
Eason Huang
Eason Huang
TechNet Community Support

Skype calls are getting failed through the application

Hi,
I keep trying to call my friend on skype and I've tried everything from reinstalling skype to power-cycling my router, but The Skype call keeps failing! If you could, help would be greatly appreciated, thanks.

Added this after looking at one of the forum entry.
restServiceAdapter.addRequestProperty("Accept", "application/json; charset=UTF-8");
And issue is resolved. Documentation update in sample code would help others.
Edited by: Chandresh on Sep 4, 2012 2:36 PM

Calls are not getting thru in Cisco voice GW for a particular Number

Cisco gateway is connecte to a PBX with an Qsig interface, for a particualr destination number the calls are not gettin estabilished.
the output of the Q931 debug :
Aug 16 16:17:46.145: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x7E05
        Bearer Capability i = 0x8090A2
                Standard = CCITT
                Transfer Capability = Speech
                Transfer Mode = Circuit
                Transfer Rate = 64 kbit/s
        Channel ID i = 0xA98396
                Exclusive, Channel 22
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
        Facility i = 0x9FAA068001008201008B0100A11D0202010002010080144E455453202
F204C4F4E472044495354414E4345
        Calling Party Number i = 0x2183, '8168911010'
                Plan:ISDN, Type:National
        Called Party Number i = 0x89, '18553808521'
                Plan:Private, Type:Unknown
        Sending Complete
Aug 16 16:17:46.149: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0xF
E05
        Channel ID i = 0xA98396
                Exclusive, Channel 22
Aug 16 16:17:55.709: ISDN Se0/0/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x
FE05
        Cause i = 0x80BF - Service/option not available, unspecified
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x7E0
5
Aug 16 16:17:55.741: ISDN Se0/0/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref =
0xFE05
The Qsig and dial-peer configration :
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-qsig
isdn overlap-receiving
isdn incoming-voice voice
isdn send-alerting
no cdp enable
dial-peer voice 1 voip
description To CBTS GK
destination-pattern +1T
signaling forward rawmsg
session protocol sipv2
session target ipv4:10.9.5.10
session transport tcp
voice-class codec 1
dtmf-relay rtp-nte
no vad

Hi Raj,
My name is Edson Pineiro, I understand that your problem description is in regards to failed incoming calls from a qsig trunk.
According to the received q931 setup message I can see the called party number is 18553808521 and as so the gateway should route the dnis based on the best match in destination-pattern. My first suggestion would be to ensure your outgoing dial-peers has a matching destination-pattern that matches the dialed number, for example:
dial-peer voice 1 voip
destination-pattern 1T
The T is a wild card for any digit any length
Or you can be very specific.
dial-peer voice 1 voip
destinaton-pattern 18553808521
The next suggestion would be to ensure that your incoming pots dial-peers contains 'direct-inward dial'. This is so that you don't receive secondary dial tone when dialing in, which I don't think is happening here.
Another suggestion would be to remove 'isdn overlap-receiving' from interface serial 0/0/0:23. Reason being is that the DNIS received is enbloc and not overlapping. You can clearly see that the complete e164 number is received within the setup and no further digits are needed.
But overall the disconnect cause code is 0x80BF the 80 portion is related to the source of the disconnect which is the router and BF "Service/option not available, unspecified" which is described as:
The network or remote equipment cannot provide the service option that the user requests, due to an unspecified reason. A subscription problem can cause this issue.
Any ways seems like the router does not support the protocol or type of message included in the Setup. After decoding one of the facility message:
        Facility i = 0x9FAA068001008201008B0100A16E0202070102011530650201010A010
1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
decode -->
Facility IE first byte (protocol profile): 0x9f(Network Extentions), depends on Network Protocol Profile
**Note:
**0x91/0x9f both be used by older qsig spec, including:
**ISO 11582:1995, ETSI 300 239:1993/1995
**newer qsig spec use 0x9f only, including:
**ISO 11582:1995/Cor.1:1999, ECMA 165(4th), ETSI 300 239:2003
**see CSCeb58118 for CCM compatibility issue
NetworkFacilityExtension ::= {
sourceEntity: 0
destinationEntity: 0
NetworkProtocolProfile not present
APDU is a ROSE
0
DivertingLegInformation2Invoke ::= {
invokeID: 1793
operationValue: 21
argument: DivertingLegInformation2Arg ::= {
diversionCounter: 1
diversionReason: 1
originalDiversionReason: 1
divertingNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
originalCalledNr: PrivatePartyNumber ::= {
privateTypeOfNumber: 2
privateNumberDigits: 50005998
redirectingName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
originalCalledName: 54 45 4C 45 43 4F 4D 20 57 4F 52 4B 52 4F 4F 4D
Looks like this is a redirected call (call forward or transfer), the redireted number is "50005998" and the other end of the PRI maybe attempting to do either a 2 B channel transfer or B channel optimization, which is not supported certain gateways or needs the use of a tcl scripts. Any ways is it possible to confirm if such features are enabled on the other end of the qsig trunk? and what the number 50005998 is assigned too. This may warrant a TAC case.
However please ensure your carry through the first three configuration changes before looking at the possible bad facility message.
Here are some good documents on ISDN, IOS dial-peers and call legs:
Understanding debug isdn q931 Disconnect Cause Codes
http://www.cisco.com/en/US/tech/tk801/tk379/technologies_tech_note09186a008012e95f.shtml
Configuring Telephony Call-Redirect Features
Two B-Channel Transfer
http://www.cisco.com/en/US/docs/ios/voice/ivr/pre12.3_14_t/configuration/guide/ivrapp.pdf
Understanding Dial Peers and Call Legs on Cisco IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010ae1c.shtml
Understanding Direct-Inward-Dial (DID) on IOS Voice Digital (T1/E1) Interfaces
http://www.cisco.com/en/US/partner/tech/tk652/tk653/technologies_tech_note09186a00801142f8.shtml
Understanding Inbound and Outbound Dial Peers Matching on IOS Platforms
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml#prereq
Voice Translation Rules
http://www.cisco.com/en/US/partner/tech/tk652/tk90/technologies_tech_note09186a0080325e8e.shtml
Let me know how you go.
Thanks again for asking the tuff questions.
Cheers
Edson

Report on transferred calls

I need to report on transferred calls. Agents are using standard CAD and selecting the transfer icon. Within the transfer form, they use the softphone to dial various 800 numbers. I need to create a report that I can group by 800 numbers called using the CAD transfer option. Does this come from the call manager? We recently implemented CVP. Are there any CVP reports that would provide this information?

Hi Ryan
I'm afraid there isn't one - only the CSQ Activity reports and the CSQ Service Level Priority Summary reports seem to show the service level info, and those are summary reports rather than reports that show call lists.
You would need to get a custom report put together for this... it's an Informix SQL procedure-writing and Crystal Reports exercise to do that.
Regards
Aaron

The Cluster Service function call 'ClusterResourceControl' failed with error code '1008(An attempt was made to reference a token that does not exist.)' while verifying the file path. Verify that your failover cluster is configured properly.

I am experiencing this error with one of our cluster environment. Can anyone help me in this issue.
The Cluster Service function call 'ClusterResourceControl' failed with error code '1008(An attempt was made to reference a token that does not exist.)' while verifying the file path. Verify that your failover cluster is configured properly.
Thanks,
Venu S.
Venugopal S ----------------------------------------------------------- Please click the Mark as Answer button if a post solves your problem!

Hi Venu S,
Based on my research, you might encounter a known issue, please try the hotfix in this KB:
http://support.microsoft.com/kb/928385
Meanwhile since there is less information about this issue, before further investigation, please provide us the following information:
The version of Windows Server you are using
The result of SELECT @@VERSION
The scenario when you get this error
If anything is unclear, please let me know.
Regards,
Tom Li

AutoAttendant not transferring calls in CUE 8.6 to CME 9.1

**Issue is calls are not transferring once extension or directory is pressed on the phone.
I upgraded our 2811 CME 7.1/CUE 7.0 router to 2911 15.2 CME 9.1/CUE8.6. It has CME and V/k9 license enabled. I uploaded the configuration used on the 2811 to the 2911. I added the ip addresses in the ip trusted list. Calls come in and out if I point the translation list to the phone and not to the auto attendant pilot number. Incase if it was toll fraud issue I disabled it but still a no go.
Everything worked fine on the 2811. I can't understand why it would not work on the 2911 15.2. Has anyone had any issues when upgrading from 12.4 to 15.x with auto attendant not transferring calls?
I ran a trace on the Cue when placed a call to extension 114:
20.10.10.1- CME interface address
20.10.10.5- CUE interface address
659 09/12 11:43:39.418 ACCN SIPS 0 Call.transferFailed(114, RESOURCE_NOT_ACKNOWLEDGING) SIPCallContact[id=33,type=Cisco SIP Call,implId=5738D3Dse-20-10-10-5# [email protected],active=true,state=CALL_ANSWERED,inbound=true,handled=false,locale=en_US

Ok I'm starting to think its a dtmf issue. I checked to see if I call from internal to the AA and dial an extension if it works but the same issue. I ran the debug voip ccapi inout and I see consume mask is not set. What would that indicate? Could that be the problem?
This is right when I dialed extension 114
2811-TEST#
001288: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001289: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001290: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001291: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001292: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001293: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001294: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001295: Sep 16 14:59:23.198: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
001296: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
2811-TEST#
001297: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
   Check DTMF relay digit begin for 3way conf
001298: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
   Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
001299: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
   Check DTMF relay digit end for 3way conf
2811-TEST#
At this point theres silence on the phone I see this message:
2811-TEST#
001300: Sep 16 14:59:28.914: //55/D62BA3D180C8/CCAPI/ccGenerateToneInfo:
   Stop Tone On Digit=FALSE, Tone=Null,
   Tone Direction=Sum Network, Params=0x0, Call Id=55
001301: Sep 16 14:59:28.918: //56/D62BA3D180C8/CCAPI/cc_api_call_feature:
   Feature Type=50, Interface=0x49FC2B80, Call Id=56
2811-TEST#
2811-TEST#
At this point I get the message "the phone number you are trying to reach" then the call disconnects
2811-TEST#
001242: Sep 16 14:48:57.719: %VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 52 GUID=5D21AC143CE711E480BCEEC6B624839
Also I checked show voice iec description:
2811-TEST#show voice iec description 1.1.129.7.67.0
    IEC Version: 1
    Entity: 1 (Gateway)
    Category: 129 (Call setup timeout)
    Subsystem: 7 (SIP)
    Error: 67 (ACK wait timeout)
    Diagnostic Code: 0
2811-TEST#

Issue with SPA525g registation and FXO port call calls are not disconnecting properly

Hi,
I have a UC540 and updated it to the latest IOS version with the latest firmware to my phones and i am having registration problems with SPA525g IP Phones. I updated the firmware of the phones as well and create manual tftp bindings with but still it is not registering. I run a couple of debugs (debug tftp events and debug ephone registration) I can see from the logs and in the phone that it is taking the proper VLAN and being discovered via CDP and being pointed to the TFTP server and still wont register. I can see that it is also taking its own .cnf file properly then the output sccp token regected invalid devices error is shown I have a SPA502G and it is working fine. Also there is a previous issue that all the voice port are shown as engage or offhook even the calls are disconnected thus make the main PSTN number busy am based in UAE and our service provider is etisalat I have check with them about the proper disconnection values but still it the same. That's why I have arrived in the conclusion to just update everything including the IOS and the phones firmware. I have put my config in this post, I am also trying to take the CCNA Voice exam on the 2nd week of april and I think that if i don't know how fix this issue for our customer then I would probably fail that exam. any suggestion and help is greatly appreciated cisco experts.
! 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?

First call attempt fails with 12000; reason="Routes available for this request but no available gateway at this point"

We've installed a new Audiocodes Mediant 1000B gateway for our customer. They only have about 6 users enabled for Enterprise voice and using Lync for all calls. They have an intermittent problem whereby the first call attempt to a number on the
PSTN fails with 12000; reason="Routes available for this request but no available gateway at this point". There is only one PSTN gateway installed. All routes point to this gateway. What I found initially is that the calls
were failing after 10 seconds which is the default "failovertimeout" in the OutboundRouting.exe.config. I found this post http://voipnorm.blogspot.co.uk/2012/06/lync-2010-gateway-timeout-call-failures.html and
changed the value to 20 seconds. Subsequent failures failed after 20 seconds (the new value). The interesting thing is that the second attempt even a couple of seconds later succeeds. My Lync server event log has 46046 "A call to a PSTN
number failed due to non availability of gateways." for the failed call and 46047 "A PBX gateway is now responding to requests after some failures." for the successful call moments later.
My environment is Lync 2010. A single enterprise edition Front End with collocated mediation. The server is virtual and in a different physical location to the gateway. The two sites are connected via a LES1000. The RTT between the
sites is very low so it isn't necessarily networking.
In the Syslog output on the Audiocodes we don't see the call even reach the gateway. It's likely that Lync has simply marked the gateway as down and doesn't route the first call. Then it wakes up and marks it as up and routes the next call.
Update wise I'm on 4.0.7577.183, 199 and 217 for those that are up to date and the only components that are behind say that 223 is available. Those being Core Components, Lync Server, Conferencing Server and Web Components.
As I said, this is intermittent, apparently doesn't happen for every user (which I don't buy), but is easily replicated on request.
I definitely think changing the failovertimeout value has reduced the number of failures. But realistically I don't want users sitting there for 20 seconds before their call fails. Or 19 seconds for the call to route.
I've found a few posts on this and similar issues. I don't get the 25051 or 25052 errors.
Any help gratefully appreciated.
Regards
Randy Chapman
Best Regards Randy Chapman

Try to change the value of Failovertimeout to 1000.
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SQL Server 2014 - calling sp_setapprole failing

The corporation I work at has decided to start using the application role functionality in SQL Server. We are able to successfully implement the app roles into the application (PowerBuilder 12.1) when running against a SQL Server 2012 database,
however the same code when running against a SQL Server 2014 database fails.
Because PB execute everything from within a transaction we had to do some "interesting" code to get the approles working. Here is the SQL that is being passed into the DB:
SET QUOTED_IDENTIFIER OFF
SET IMPLICIT_TRANSACTIONS OFF
SET ANSI_PADDING ON
create table "<generated temp table name>" (id int identity( 1, 1) not null, cookie varbinary(8000) not null )
declare @cookie1 varbinary( 8000) exec sp_setapprole @rolename = '"+ <rolename>+"', @password = '"+<password>+"', @fCreateCookie = 1, @cookie = @cookie1 output insert "+<generated temp
table name>+"( cookie) values( @cookie1)
So what happens is the approle is being called and the cookie stored in a temp table on the database. This is because of the translation between SQL Server and PB with the Varbinary cookie that is created for the approle. If brought into PB it gets
messed up and the unset approle doesn't work.
As stated, the code above WORKS in SQL Server 2012, not in 2014. I run the profile trace on the server and I can see the temp table being created, the sp_setapprole proc being called and the statements inside being executed, but once hitting the function
setuser we get the error that the approle does no exist or the password is invalid. I've been able to trace through the password being sent in and it's the same as that PB is sending across so it's not a corruption issues.
Running the SQL through SS Management Studio works. The temp table created and the approle is set. Additionally we have .Net applications using the same user/password, and retrieving a cookie, and they work as well.
Does anyone know if there has been changes to the way approles are handled in 2014, or if there was a change to the setuser functionality that could be causing the approle to fail?

The call "SET QUOTED_IDENTIFIER OFF" was an attempt to get the DB working the same way our SQL Server 2012 database is set-up. I've since removed it
from the code. Also each of the calls are executed separately:
ls_sql = "SET IMPLICIT_TRANSACTIONS OFF"
EXECUTE IMMEDIATE :ls_sql USING atrTrans;
IF atrTrans.SqlCode <> 0 THEN
guMsg_handler.uf_message(TRUE, &
"Unable to connect to the database.~r~n" + &
atrTrans.sqlerrtext)
RETURN false
END IF
ls_sql = "SET ANSI_PADDING ON"
EXECUTE IMMEDIATE :ls_sql USING atrTrans ;
IF atrTrans.SqlCode <> 0 THEN
guMsg_handler.uf_message(TRUE, &
"Unable to connect to the database.~r~n" + &
atrTrans.sqlerrtext)
RETURN false
END IF
ls_sql = "create table "+atrTrans.isTempTableName+"( id int identity( 1, 1) not null, cookie varbinary(8000) not null ) "
EXECUTE IMMEDIATE :ls_sql USING atrTrans ;
IF atrTrans.SqlCode <> 0 THEN
guMsg_handler.uf_message(TRUE, &
"Unable to connect to the database.~r~n" + &
atrTrans.sqlerrtext)
RETURN false
END IF
ls_sql = "declare @cookie1 varbinary( 8000), @password1 sysname = '" + ls_appRollPass + "' exec sp_setapprole @rolename = '"+ ls_AppRoleName +"', @password = @password1, @fCreateCookie
= 1, @cookie = @cookie1 output insert "+atrTrans.isTempTableName+"( cookie) values( @cookie1) "
EXECUTE IMMEDIATE :ls_sql USING atrTrans ;
IF atrTrans.SqlCode <> 0 THEN
guMsg_handler.uf_message(TRUE, &
"Unable to connect to the database.~r~n" + &
atrTrans.sqlerrtext)
RETURN false
END IF
atrTrans.AutoCommit = FALSE
ls_sql = "SET IMPLICIT_TRANSACTIONS ON"
EXECUTE IMMEDIATE :ls_sql USING atrTrans ;
ls_sql = "SET ANSI_PADDING OFF"
EXECUTE IMMEDIATE :ls_sql USING atrTrans ;
commit using atrTrans ;
EDIT: Note that the passing of the password as a variable was only for testing/validation to ensure that the password matched what was being sent from PB and NOT intended to be there in the actual implementation.

Transfered calls are failing

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