Cisco IVR

hi, i am using a AS5300 and currently exploring the IVR scripts. Here is my current config
aaa new-model
aaa authentication login h323 group radius
aaa authorization exec h323 group radius
aaa accounting connection h323 start-stop group radius
aaa session-id common
gw-accounting aaa
method voip
attribute acct-session-id overloaded
attribute h323-remote-id resolved
interface Ethernet0
description WAN-connection Duplex-not-support on E0
ip address 192.168.10.240 255.255.255.0 h323-gateway voip interface
h323-gateway voip h323-id ivr-testing
radius-server host 172.16.16.25 auth-port 1812 acct-port 1813
radius-server key xxx
radius-server vsa send accounting
radius-server vsa send authentication
call application voice ivr-testing tftp://172.16.16.30/TCLware/clid_col_npw_3_cli.1.1.0.tcl
call application voice ivr-testing language 0 en
call application voice ivr-testing set-location en 0 tftp://172.16.16.30/TCLware/prompts/en
dial-peer voice 1083 voip
description IVR Testing
application ivr-testing
destination-pattern 02070788133
session protocol sipv2
session target ipv4:172.16.16.30 dtmf-relay rtp-nte
codec g711ulaw
dial-peer voice 1082 pots
description IVR Testing - Do not Remove application ivr-testing
destination-pattern 44070788133
port 2:D
My queries about this is that, when i dial the DID using Cisco 7960 ip phone my attempts wont able to reach the router. Hence using an analog phone got an error no route to destination. Using my analog phone, it is connected to another gateway equipment. and my gateway is pointed to the cisco as 5300. I need some expert advise. Thanks

wait, wait...
Your objective than gives negative answer to second question because if you have analog phone connected to Quintum that call still goes over IP and comes on VOIP dial-peer to AS5300, not via PSTN and POTS dial-peer?
I am not sure if you can accept voip call and forward it to IVR because both calls are on voip side...
As I understand, IVR is not a problem in your case, call routing is?
i would suggest that first you try with simple call scenario without IVR just to be sure that calls are coming to AS5350.
Still, I am sending you a document where yoo can find more information about configuring IVR.
Cheers!

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Hi All,
All i need is to write a script in UCCX premium with below requirement.
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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
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DEFAULT Default system session application
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Call Treatment Action Application - Version 1.0.
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TCL Script Version 1.1 supported.
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Hi,
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Cheers.

This may help you find some command to reduce it
http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_command_reference_chapter09186a00801a7f3d.html

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George

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<?xml version="1.0" ?>
http://yourwebserver or flash/hello_world.au"/>
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Hi
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Regards
Aaron
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Sent from Cisco Technical Support iPad App

Thanks for ur reply...
yes right..I agree ur opinion..I was already requested to SE/AM, even PSS..
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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
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1800101A111A00FA50D0A010212083530303035393938A211A00FA50D0A010212083530303035393
938A312801054454C45434F4D20574F524B524F4F4DA412801054454C45434F4D20574F524B524F4
F4DA50C06062B0C02FF373730020500
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F204C4F4E472044495354414E4345
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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
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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
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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 'the number of calls' on an IVR Option menu

Dear NetPro gurus,
My customer has an IVR on UCCX which works fine. However, they have recently add in an 'Option 6' where when PSTN caller press on it, it will redirect the call out to an 1800 number onto the PSTN.
The call connects perfectly fine. What my customer wants is to have a report on how many PSTN caller press on this newly created Option 6 of the IVR.
For all the other 5 IVR options, they all have an individual CSQ assigned to them and there is a 'Set Enterprise Call Info' step to send the 'options selected' to UCCX.
However, for this Option 6, since there is no CSQ (because the call gets redirect to a PSTN number), is there any way i can script it so that it can be tally and identified by Cisco Historical Reports too?
I have spent a lot of time on this already so would appreciated if anyone can shed some light on this.
Cheers,
Hunt

Hi Hunt
There are quite a few posts on the subject on the forums.
Basically you can do one of these things:
1) Run an 'application summary' report - typically the count of 'flow out' calls will be the number of calls sent to your 1800 number, if this is the only non-CSQ destination that your script sends calls to (e.g. if there are no other external numbers, voicemail, or internal DNs that you send the calls to other than via a CSQ).
2) Run a 'custom call variables' report and then count up the number of rows that have the info that you set on 'set enterprise call info' when the call was routed externally
3) If those aren't good enough, then you can produce a custom report to show the data in the format the customer wants. This would require some good SQL and Crystal skills...
Regards
Aaron HarrisonPrincipal Engineer at Logicalis UK
Please rate helpful posts...

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