10-15-2006 04:35 AM - edited 03-13-2019 03:24 PM
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.
10-15-2006 08:14 AM
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.
10-15-2006 10:38 AM
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?
10-15-2006 10:51 AM
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.
10-15-2006 11:39 AM
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
10-20-2006 10:03 PM
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.
10-21-2006 01:42 AM
10-22-2006 08:37 AM
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>5512041843>
From: <5557996297>;tag=738934c8-13c4-453baa94-54d1ad7a-7935557996297>
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
10-22-2006 01:29 PM
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=929b93e039d4c092fd8ab883667433712>
To: <964812530>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>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
10-23-2006 10:52 AM
Hola Santi
Mandame un correo a mi direcci?n para que te pueda enviar la utilidad.
gogasca arroba cisco punto com
10-23-2006 11:34 AM
ya te he mandado un correo, espero que pueda solucionar mi problema, muchas gracias por la colaboraci?n.
10-15-2006 08:17 AM
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
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