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

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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?
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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]
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Your help is highly appreciated.
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Restart your phone by holding the power and home button until the Apple logo appears. If that does not solve the issue contact your cell provider so they can resolve the issue.

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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
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Hi Irfan,
Thanks for the reply.
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Fatal NI connect error 12537, connecting to:
(LOCAL=NO)
VERSION INFORMATION:
TNS for Linux: Version 11.1.0.7.0 - Production
Oracle Bequeath NT Protocol Adapter for Linux: Version 11.1.0.7.0 - Production
TCP/IP NT Protocol Adapter for Linux: Version 11.1.0.7.0 - Production
Time: 15-MAY-2012 16:49:15
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ns main err code: 12537
TNS-12537: TNS:connection closed
ns secondary err code: 12560
nt main err code: 0
nt secondary err code: 0
nt OS err code: 0
ORA-609 : opiodr aborting process unknown ospid (18807_47295439087424)
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A couple of thoughts come immediately to mind:
1. When I read ... "Tracing not turned on" ... I wonder to myself ... why not turn on tracing?
2. When I read ... "Version 11.1.0.7.0" ... I wonder to myself ... why not apply all of the patches Oracle has created in the last 3 years and see if having a fully patched version addresses the issue?
3. When I read ... "parallel query server died" ... I wonder whether you have gone to support.oracle.com and looked up the causes and solutions for Parallel Query Server dying?
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Morning sandyab60,
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Hi, Kenny1955, and welcome to the Community,
Sounds like a call routing issue. Please send a private message to me with as much detail as possible listing the numbers or exchanges from which people may have reported to you that their calls did not get through to your Skype Number as well as the carriers in question, if known (mobile or wireless network as well as fixed line).
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Was your question answered? Please click on the Accept as a Solution link so everyone can quickly find what works! Like a post or want to say, "Thank You" - ?? Click on the Kudos button!
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My calls are being blocked from calling our office.

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Hello @susankip
I’m sorry you’re not able to dial out to your office.
It sounds like there is definitely a routing or dialing pattern issue going on.
To better assist with this, please contact us as soon as possible.
You are welcome to use our live chat option as well.
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Calls not coming through

Hi have the subscription for unlimited calling to both the U.S and Canada. My calls are coming through from U.S landlines and mobiles but when someone from Canada tries calling all they get is a beeping noise. Can someone please help me.

kittycopper wrote:
Hi have the subscription for unlimited calling to both the U.S and Canada. My calls are coming through from U.S landlines and mobiles but when someone from Canada tries calling all they get is a beeping noise. Can someone please help me.
If you have the Unlimited USA & Canada subscription, you should be able to call telephones in both countries. If someone is trying to call you, are they calling you directly to your Skype account name? Did you also purchase an Online number , so people without Skype can call your Skype account from a telephone? Purchasing that subscription by itself does not provide a way to receive calls from telephones.
Patrick
Location/Ubicacion: Arizona USA
Time Zone/Hora Local: UTC/GMT -7
If this message has adequately addressed your issue, please click on the “Accept as Solution” button. If you found a post useful then please "Give Kudos" at the bottom of my post, so that this information can benefit others.
Si esto mensaje le ha ayudado, por favor haga clic en "Aceptar como solución". Si encuentra un mensaje útil, por favor "Da Kudos" al final del mensaje, por lo que esta información puede beneficiar a otros.
I am not a Skype employee. No soy un empleado de Skype.

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

Some of my calls are not coming through but going straight to voicemail....anything I can do?

Some calls are coming in and some aren't. Is there something I should do to my phone or is this a network issue?

Try resetting your network settings.
Settings > General > Reset > Reset Network Settings
If this doesn't resolve it, contact your wireless carrier for assistance, as the problem is most likely with the network, not your phone.

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

Iphone 4S; Phone & audio piece; Incoming calls are NOT routed automatically??

When I dial out; no problem the calls automatically goes through ear pice.
When I get incomming calls the call goes to phone, not even to the speaker in in the phone. So I have to
1.Slide the bar (then first realizing (again) dam: I hear nothing)
2. Choose the box for speaker options
3. Choose my blue ant piece..
By then some callers think I am not picking up or there is somthing wrong..!
My 3 colleaguers have same phone and no such problems. We are at a loss and I could not find reply in community groups. There was note about Iphone 5 and not correcting to Car speaker automatically. This was noted as a softwre issue Apple might address in next version. But since my colleagues do not have problem with their Iphone it must be either my settings or a weird fault with my phone.
One of my collegues have same ear pice as me, so the issueis not that either.
Is there some weird setting I need to fix to correct the problem?
Thank you

Are you able to make calls? I would try a couple different things -
1. Try doing a double hard reset - Press and hold home button and the sleep wake button at the same time until the screen displays the slide to shut down red button. Continue pressing and holding both the buttons until the screen goes black and then the apple logo flashes up. Continue pressing and holding both the home and sleep wake button beyond this point until the screen goes black again. At this point, let go of the home button and press once the sleep wake button as you would to start your phone. See if this double hard reset helps clear out any cache.
2. If this fails, I would try restoring the phone as a new phone in iTunes. Backup your phone before this so that you dont lose your contacts, etc. You can do a couple test calls once the firmware is installed before setting up the phone, data, apps, etc. If it looks good, try restoring your phone from an earlier backup.
If everything fails, call AppleCare or see a Genius in the store.

Support Line - Call Forward All through Hunt Pilot to cell phone

I know I may be beating a dead horse, but I've found so much help in these forums that I figured I'd give it a shot.
We have a Cisco Unified Communications Manager Business Edition and we are running into a configuration issue. What we would like to do is to create an emergency support line for our IT staff. The way we would like this to function is that a caller dials a number, CM accepts the call and dials out to our IT staff's cell phone numbers in a round-robin fashion. To avoid the caller getting dumped into the IT staff's voice mail on their cells, we would like the staff member to have to dial a number to accept the call. If there is no answer, the call rolls to the next cell number. If no one is available, the caller should be directed to Unity Connection to leave a message. Unity then will send out text and email messages to the support staff.
I know that we can use Unity to perform an assisted transfer, which will require the user to press "1" to accept the call, and we are able to get Unity to send out the notifications (text and email) when a voice message is left. The issue is with Call Manager making the outbound calls to the cell phones.
What we have attempted is to setup DNs that call forward all to the users cell numbers. These DNs have been added to a Line Group, which has a Hunt Pilot attached to it. Any time this pilot is called, we get a reorder. Using the DNA, we see that "no device is associated with the DN", which is under the DN for the first users cell forward. If we add that DN as a second line to that users IP Phone (7940), then the call into the Hunt Pilot rings that line on the IP phone, not the CFA to the cell phone.
After weeks of digging around, it seems as though CFA in a Hunt is just not possible. My boss wants official word from Cisco about this, but TAC doesn't seem to want to help due to service contract issues (which blows my mind, as we have opened several cases in the past two months for configuration related issues). Our Business Edition came with Contact Center Xpress, although we do not have the resources available to install it. If CCX will carry out this task, that might be enough to push management into getting another server to support this, but without being able to play around with it, I don't know.
If anyone has any suggestions on how to make this work, I would GREATLY appreciate the help!
Thanks in advance,
-Geoff

Hi Geoff,
Always interesting isn't it :)
Call Forward settings on individual Hunt member phones are ignored when presented a call via the Hunt feature. Here is a clip;
The concept of hunting differs from that of call forwarding. Hunting allows Cisco CallManager to extend a call to one or more lists of numbers, where each such list can specify a hunting order that is chosen from a fixed set of algorithms. When a call extends to a hunt party from these lists and the party fails to answer or is busy, hunting resumes with the next hunt party. (The next hunt party varies depending on the current hunt algorithm.) Hunting thus ignores the Call Forward No Answer (CFNA) or Call Forward Busy (CFB) settings for the attempted party. This also applies to CFWD ALL settings.
From this doc;
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a00803edabe.html#wp107892
So, any Forwarding settings have to be done on the Hunt Pilot itself. These Destination settings (on the Hunt Pilot) need to be configured to go to the Unity Voicemail Pilot # or perhaps this is where you Forward out to the Cell #;
Hunt Forward Settings
Forward Hunt No Answer - When the call that is distributed through the hunt list is not answered in a specific time, this field specifies how to forward the call.
Destination This setting indicates the directory number to which calls are forwarded. (This can be the Directory Number of the Unity VM Pilot)
Forward Hunt Busy - When the call that is distributed through the hunt list is busy in a specific time, this field specifies how to forward the call.
Destination This setting indicates the directory number to which calls are forwarded. (This can be the Directory Number of the Unity VM Pilot)
Maximum Hunt Timer - Enter a value (in seconds) that specifies the maximum time for hunting.(Used in conjunction with Forward Hunt Busy)
Maybe you could leverage these Unity Connection configs to achieve your desired results. These will ensure that the Message is not left unattended;
Cascading Message Notification
Cascading message notification allows you to send notifications to a widening circle of recipients. Cisco Unity Connection continues to send notifications according to the devices you selected until the message has been saved or deleted by a recipient.
For example, to create a cascade of message notifications for your Technical Support department,
Chaining Message Notification
Message notification can be set to "chain" to a series of notification devices if an attempt to send notification to the first selected device fails. The definition of failure to a notification device is based on the options you select for retrying a device that is not answered or is busy.
http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac/guide/2xcucmac040.html#wp1132107
http://www.cisco.com/en/US/docs/voice_ip_comm/connection/2x/user_mac_cmbe/guide/6xcucmbemac040.html#wp1132107
Hope this helps!
Rob

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?

PCS calls are failing through "SendToVRU" node.

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