autoattendant configure in cme

[sarenos2006]

I am trying to configure an autoattendant in cme 3.3.

My tcl script is called app-b-acd-aa-2.1.0.0.tcl

My config is attached.

But autoatt it's not running, when I have put these commands you can see below I have had warnings:

Router(config-app-param)#param operator 6001

Warning: parameter operator has not been registered under autoatt namespace

Router(config-app-param)#param aa-pilot 6060

Warning: parameter aa-pilot has not been registered under autoatt namespace

Is necessary to create a new ephone-dn with number 1000? Why I have this warnings?

Please you help will be great. I need information for continuing investigating why is not working

Thanks in advanced.

[gogasca]

Those are warnings, can be ignored.

Ok aa-pilot will be used as the called number for the incoming-dial peer that will be reciving the call.

For example:

Your telco delivers a 4 digit number for incoming calls, you main number is 1000.

So you configure the incoming called number command with 1000, and the param aa-pilot 1000

For Operator param operator, its the extension that will ring when people dial 0 if I remmeber correctly or doenst enter a number.

Dont dial from an IP Phone it wont work, dial from either the PSTN or you remote SIP network.

[sarenos2006]

I have done your recommendations.

My telco delivers me the number 964812530

so I have put it in incoming called number and aa-pilot.

In param operator I have put 200, that is the operator extension.

But the system continues without running correctly.

As you can see in the configuration I have translation rules for incoming calls, I have disabled it and it's the same.

How can I do debugs? Can you see my config file and tell me your oppinion?

[gogasca]

Ok please do a debug voice ccapi inout

debug ccsip all.

You Telco delivers the number 964812530 using which protocol?

I see that you configure: paramspace english location flash:

Please confirm that you have all the .au files there.

[sarenos2006]

My telco is a sip provider ( sip protocol ), so I have configured a sip account in the router with the username is 964812530 and pass xxxxxx

so the public number is 964812530

these are the files I have in the flash:

Router#sh flash:

System flash directory:

File Length Name/status

1 25886372 c2600-jsx-mz.124-1

2 1022976 cme-b-acd-2.1.0.0.tar

3 33949 app-b-acd-aa-2.1.0.0.tcl

4 83291 en_bacd_disconnect.au

5 37952 en_bacd_invalidoption.au

6 123446 en_bacd_options_menu.au

7 75650 en_bacd_allagentsbusy.au

8 63055 en_bacd_enter_dest.au

9 496521 en_bacd_music_on_hold.au

10 42978 en_bacd_welcome.au

[27866844 bytes used, 5687584 available, 33554428 total]

32768K bytes of processor board System flash (Read/Write)

When I try to call 964812530 from the PSTN (the number from I'm trying the calls is 667433712 ) this is the

debug voice ccapi inout :

Router#debug voice ccapi inout

voip ccapi inout debugging is on

Router#

*Mar 27 20:54:10.988: //17/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:

Call Entry Is Not Found

*Mar 27 20:54:10.992: //-1/9F6EA05E801C/CCAPI/cc_api_display_ie_subfields:

cc_api_call_setup_ind_common:

cisco-username=667433712

----- ccCallInfo IE subfields -----

cisco-ani=667433712

cisco-anitype=0

cisco-aniplan=0

cisco-anipi=0

cisco-anisi=0

dest=964812530

cisco-desttype=0

cisco-destplan=0

cisco-rdie=FFFFFFFF

cisco-rdn=

cisco-rdntype=0

cisco-rdnplan=0

cisco-rdnpi=0

cisco-rdnsi=0

cisco-redirectreason=-1

*Mar 27 20:54:10.992: //-1/9F6EA05E801C/CCAPI/cc_api_call_setup_ind_common:

Interface=0x84E54E24, Call Info(

Calling Number=667433712(TON=Unknown, NPI=Unknown, Screening=Not Screened, Pr

esentation=Allowed),

Called Number=964812530(TON=Unknown, NPI=Unknown),

Calling Translated=FALSE, Subsriber Type Str=Unknown, FinalDestinationFlag=TR

UE,

Incoming Dial-peer=1001, Progress Indication=NULL(0), Calling IE Present=TRUE

,

Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALS

E), Call Id=17

*Mar 27 20:54:10.996: //-1/9F6EA05E801C/CCAPI/ccCheckClipClir:

In: Calling Number=667433712(TON=Unknown, NPI=Unknown, Screening=Not Screened

, Presentation=Allowed)

*Mar 27 20:54:10.996: //-1/9F6EA05E801C/CCAPI/ccCheckClipClir:

Out: Calling Number=667433712(TON=Unknown, NPI=Unknown, Screening=Not Screene

d, Presentation=Allowed)

*Mar 27 20:54:10.996: //17/9F6EA05E801C/CCAPI/cc_api_call_setup_ind_common:

Set Up Event Sent;

Call Info(Calling Number=667433712(TON=Unknown, NPI=Unknown, Screening=Not Sc

reened, Presentation=Allowed),

Called Number=964812530(TON=Unknown, NPI=Unknown))

*Mar 27 20:54:11.000: //17/9F6EA05E801C/CCAPI/cc_process_call_setup_ind:

Event=0x85284678

*Mar 27 20:54:11.016: //17/9F6EA05E801C/CCAPI/ccCallSetContext:

Context=0x85BC79E4

*Mar 27 20:54:11.016: //17/9F6EA05E801C/CCAPI/cc_process_call_setup_ind:

>>>>CCAPI handed cid 17 with tag 1001 to app "_ManagedAppProcess_autoatt"

*Mar 27 20:54:11.024: //17/9F6EA05E801C/CCAPI/ccCallDisconnect:

Cause Value=16, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect C

ause=0)

*Mar 27 20:54:11.024: //17/9F6EA05E801C/CCAPI/ccCallDisconnect:

Cause Value=16, Call Entry(Responsed=TRUE, Cause Value=16)

*Mar 27 20:54:11.212: //17/9F6EA05E801C/CCAPI/cc_api_call_disconnect_done:

Disposition=0, Interface=0x84E54E24, Tag=0x0, Call Id=17,

Call Entry(Disconnect Cause=16, Voice Class Cause Code=0, Retry Count=0)

*Mar 27 20:54:11.212: //17/9F6EA05E801C/CCAPI/cc_api_call_disconnect_done:

Call Disconnect Event Sent

The debug ccsip all is attached.

And the config is attached too.

and the .tcl script is attachef too

I hope we can find a solution.

thanks in advanced

[gogasca]

Hi

I would like to know if you were able to get this up and running?

If not let us know so we can help you with your config.

[sarenos2006]

sip configuration is running perfectly. But now I'm trying to configure an autoattendant but its impossible. It doesn't works.

I have read all cisco documents and I think everything is correct but it doesn?t run.

Could you check my config?

[gogasca]

Hi Sarenos,

Your config looks good.

One last question, when you configure your CCME, to recieve a direct call from your Telco to your IP Phone...without attempting to use the TCL Script. which codec is being used. Press i button during the active this may be a codec issue.

debug ccsip messages

debug ccsip events

ter mon

You will see something like:

*Jan 4 13:44:43.342: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:5512041843@X.X.X.X:5060 SIP/2.0

Max-Forwards: 9

Session-Expires: 3600;Refresher=uac

Supported: timer

To: <5512041843>

From: <5557996297>;tag=738934c8-13c4-453baa94-54d1ad7a-793

Call-ID: 375077-3370523608-630180@protelmsw1.subnet32.net

CSeq: 1 INVITE

Via: SIP/2.0/UDP 200.76.111.52:5060;branch=e21091e183f180b783564613e29e8705

Contact: sip:5557996297@200.76.111.52:5060

Content-Type: application/sdp

Content-Length: 287

v=0

o=NexTone-MSW 1234 1161538915 IN IP4 200.76.111.53

s=sip call

c=IN IP4 200.76.111.53

t=0 0

m=audio 27074 RTP/AVP 18 4 8 17 96

a=rtpmap:18 G729/8000

a=rtpmap:4 G723/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:17 T38/8000

a=fmtp:96 0-15

a=rtpmap:96 telephone-event/8000

a=ptime:20

As you can see in the SDP message the first codec announced by my Telco in attributes field for Media is G729 and prompts are recorded in G711U.

Please confirm

[sarenos2006]

OK, I have tested it without using the tcl script and doing the call directly.

I have done a debug ccsip messages

and the result has been:

outer#debug ccsip messages

SIP Call messages tracing is enabled

Router#

*Mar 27 21:40:24.323: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Received:

INVITE sip:964812530@192.168.2.200:5060 SIP/2.0

Via: SIP/2.0/UDP 213.162.201.146:5060

Via: SIP/2.0/UDP 213.162.201.147:5060;branch=z9hG4bK929b93e0392b0db1fd8ab881

Max-Forwards: 69

From: <667433712>;tag=929b93e039d4c092fd8ab883

To: <964812530>

Call-ID: 929b93e0396c4a90fd8ab880@213.162.201.147

CSeq: 1 INVITE

User-agent: SysMaster VoIP Gateway v1.2.0

Contact:

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, INFO, REFER, SUBSCRIBE

Content-Type: application/sdp

Content-Length: 426

Record-Route: <213.162.201.146:5060>

v=0

o=- 220891596850131 1 IN IP4 213.162.201.146

s=-

c=IN IP4 213.162.201.146

t=0 0

m=audio 17018 RTP/AVP 3 0 8 18 4 99 98 97 96 101

a=rtpmap:3 GSM/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000

a=rtpmap:99 G726-16/8000

a=rtpmap:98 G726-24/8000

a=rtpmap:97 G726-32/8000

a=rtpmap:96 G726-40/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

that's true the fist codec is g729.

and the promts are in g711, so what I need to do?

Convert de promts to g729?, Create new promts in g729?

Which utility can I use for doing the conversion?

Attached is the complete debug ccsip messages

[gogasca]

Hola Santi

Mandame un correo a mi direcci?n para que te pueda enviar la utilidad.

gogasca arroba cisco punto com

[sarenos2006]

ya te he mandado un correo, espero que pueda solucionar mi problema, muchas gracias por la colaboraci?n.

[gogasca]

The error message you seeing is just a warning that although you have configured them, they will not take effect until the next reload of the box or application (call

application voice load