UC540 AA problem

[Velin Tsekov]

Dear All,

I have the following configuration, concerning Auto Attendand, running on UC540:

dial-peer voice 502 voip

description $AUTO ATTENDAND$

destination-pattern 298

b2bua   

session protocol sipv2

session target ipv4:10.3.3.162

voice-class sip outbound-proxy ipv4:10.3.3.162 

dtmf-relay sip-notify

codec g711ulaw

no vad

Unity Express:

ccn trigger sip phonenumber 298

application "autoattendant"

enabled

maxsessions 6

end trigger

ccn application autoattendant aa

description "autoattendant"

enabled

maxsessions 6

script "aa.aef"

parameter "busClosedPrompt" "AABusinessClosed.wav"

parameter "holidayPrompt" "AAHolidayPrompt.wav"

parameter "welcomePrompt" "AAWelcome.wav"

parameter "disconnectAfterMenu" "false"

parameter "dialByFirstName" "false"

parameter "allowExternalTransfers" "false"

parameter "MaxRetry" "3"

parameter "dialByExtnAnytime" "false"

parameter "busOpenPrompt" "AABusinessOpen.wav"

parameter "businessSchedule" "systemschedule"

parameter "dialByExtnAnytimeInputLength" "4"

parameter "operExtn" "206"

end application

When I call 298 from internal number, I hear the prompt, select 0 or 1 and everything is OK. When i try from outside (mobile) number I hear the prompt, select 0 or 1 (dtmf work) and I don't here anythin after that. I receive this error in the debug message:

002069: Jun  6 18:17:40 EEST: //1487/F1D1F58B8684/CCAPI/cc_api_get_ssCTreRoutingNotSupported:

I said that the dtmf works, because if i press "1" I hear the prompt enter the phone number you trying to reach.

Any tips?

[mtempro · Accepted solution]

Hello Velin,

Your AA dial-peer appears to have the correct configuration even though changing the cue ip address isn't recommended.  What kind of incoming trunk are you using, Sip, PRI, FXO? It would be helpful to take a look at your incoming dial-peer configuration.  With some sip providers, the issue may be on their end.

Regards,

Marvin

[Velin Tsekov]

I use SIP Trunk with the local provider. Here you are the incoming dial-peer config:

dial-peer voice 101 voip
description $SIP TRUNK FIXED INCOMING$
translation-profile incoming BTC-INCOMING
session protocol sipv2
session transport udp
incoming called-number 290230T
dtmf-relay rtp-nte cisco-rtp
codec g711ulaw

I installed seveal UCs for customer of ours and I don't have the problem witch i noticed here. As I sait it seems that the dtmf working properly.

[Velin Tsekov]

Fixed...

I missed some config on the voice service voip.

[mtempro]

Hi, I would recommend dropping the Cisco proprietary cisco-rtp and just use the RFC 2833 standard - dtmf-relay rtp-nte for your inband signaling.  You could also check with the SP and verify that the proxy server you're connecting to also supports 2833.

-Marvin

[mtempro]

Was it the following,

alllow-connections h323 to h323

allow-connections sip to sip

allow-connections sip to h323

allow-connections h323 to sip

Glad to see you were able to get it going.  I didn't want to recommend this without first knowing what incoming trunk type you're using.