AutoAttendant CUE - call redirect problem

apapakons · ‎05-24-2013

Recently I have installed a cme and in addition I integrated a cue to it. The problem is that when I call to the central AutoAttendant number (10 digits number) the call redirection to external numbers fails (I use customised script with with "call-redirect" parameter from aa script editor .ie call-redirect external_number). On the contrary AA can succesfully redirect all calls to internal telephone numbers. I can show you the flow of a calll with a translations:

PROVIDER MY LAN

+30xxxxxxxxxxx -------------------------------------------> CME ------------------------------------>CUE------>call redirect to external number----->fails

country code translation 10digits number 10digits number

(+30) plus 10 digits remove (+30)

number.

the dial-peers that I use are for AA are:

dial-peer voice 1002 voip

description **AA**

destination-pattern 2132144890

b2bua

session protocol sipv2

session target ipv4:"IP of CUE"

dtmf-relaysip-notify

codec g711ulaw

no vad

dial-peer voice 1003 voip

description ***incoming AA***

translation-profile incoming PSTN_INCOMING2 (removes +30 from the incoming calls from provider)

incoming called-number +302132144890

session protocol sipv2

session target ipv4:195.167.16.153:5060

incoming called-number .

voice-class codec 1

dtmf-relay rtp-nte

no vad

Also:

telephony-service

transfer-pattern 213214....

call-forward pattern 213214.... (to allow call transfers and forwards to external numbers)

voice service voip

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

Fianlly I have attached a log file of the command "debug ccsip messages" when I call the AA and trire to redirect my call.

May be I need an extra rule in dial-peer 1002, in order to add +30 to the extrenal number that the AA tries to reach and fails.The provider needs +30 (contry code) in front of each 10 digit number. I would really appreciate any help.

paolo bevilacqua · ‎05-24-2013

If provider needs country code, put country code.

apapakons · ‎05-26-2013

First of all thanks Paolo for your help. The translation rule that I used for my phones was

voice translation-rule 5

rule 1 /^$..........$$/ /+30\1/

voice translation-profile PSTN_OUTGOING

translate calling 5

So I added +30 (area code) to the calling number of an outgoing call. In the case of Auto Attendant:

a)can I use the same translation profile? Do I translate the only the calling number which is the Auto Attendant 10 digit number (how should the translation rofile look like for AA)?

b)in which dial peer should I apply the dial-peer? in dial-peer 1002?

c)In the dial peer should I apply an outgoing or incoming translation profile?

Sorry for the question but I am a little bit confused.

apapakons · ‎05-27-2013

I found a mistake in my custom aa script. I use the "call redirect" parameter to make the AA to redirect a call to an external number. I forgot to add the digit '9' in front of the external number, so an outgoing dial-peer did not matched. Generally, I use 9 to get a dial tone for phones and the I remove it with translation rules.

After this change I tried to debug the dial peers to make sure the right outgoing dial-peer is used. Gladly I found out that the outgoing dial-peer for domestic calls was used:

voice translation-rule 1

rule 1 /^48$[012345].$$/ /21321448\1/

rule 2 /^48$6[12345]$$/ /21321448\1/

rule 3 /^948$..$$/ /21321448\1/

voice translation-rule 5

rule 1 /^$21321448[789].$$/ /+30\1/

rule 2 /^$213214486[56789]$$/ /+30\1/

voice translation-rule 2

rule 1 /^9$2.........$$/ /\1/

rule 2 /^9$.......$$/ /210\1/

rule 3 /^9$100$$/ /100/

rule 4 /^9$69........$$/ /\1

voice translation-profile PSTN_OUTGOING

translate calling 5

translate called 2

translate redirect-target 1

translate redirect-called 1

dial-peer voice 2 voip

corlist outgoing uselocal

description OUTGOING PSTN - ASTIKA

translation-profile outgoing PSTN_OUTGOING

destination-pattern 921........

session protocol sipv2

session target ipv4:x.x.x.x

voice-class codec 1

dtmf-relay rtp-nte

no vad

This dial-peer works like a charm for all my phones, but it returns me a SIP ERROR 404 when the CUE AA redirects a call to an external number. The debug when AA redirects a call to an external number is:

May 27 13:03:58.048: //1420/03C74BD5ACBA/SIP/Msg/ccsipDisplayMsg:

Sent:

INVITE sip:2132144813@X.X.16.153:5060 SIP/2.0

Via: SIP/2.0/UDP X.X.71.34:5060;branch=z9hG4bK325190B

Remote-Party-ID: <>2132144809@X.X.gr>;party=calling;screen=no;privacy=off

From: <>2132144809@X.X.gr>;tag=10CEAFC4-1376

To: <2132144813>

Date: Mon, 27 May 2013 13:03:58 GMT

Call-ID: BA311B85-C60411E2-8CD8F11B-22DC6BD2@ims

namuseum_CUCME#X.X.gr

Supported: 100rel,timer,resource-priority,replaces,sdp-anat

Min-SE: 1800

Cisco-Guid: 0063392725-3322221026-2897914903-3929223243

User-Agent: Cisco-SIPGateway/IOS-12.x

Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER

CSeq: 101 INVITE

Timestamp: 1369659838

Contact: <2132144809>

Expires: 180

Allow-Events: telephone-event

Max-Forwards: 61

Alert-Info: <http://www.huawei.com/ring/>;info=pattern1

Content-T

namuseum_CUCME#undebug aype: application/sdp

Content-Disposition: session;handling=required

Content-Length: 246

v=0

o=CiscoSystemsSIP-GW-UserAgent 6555 109 IN IP4 X.X.71.34

s=SIP Call

c=IN IP4 X.X.71.34

t=0 0

m=audio 31284 RTP/AVP 8 101

c=IN IP4 X.X.71.34

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=ptime:20

May 27 13:03:58.064: //1420/03C74BD5ACBA/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP X.X.71.34:5060;branch=z9hG4bK325190

namuseum_CUCME#undebug allB

From: <>2132144809@X.X.gr>;tag=10CEAFC4-1376

To: <2132144813>

Date: Mon, 27 May 2013 13:06:25 GMT

Call-ID: BA311B85-C60411E2-8CD8F11B-22DC6BD2@X.X.gr

Timestamp: 1369659838

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Content-Length: 0

May 27 13:03:58.064: //1420/03C74BD5ACBA/SIP/Msg/ccsipDisplayMsg:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP X.X.71.34:5060;branch=z9hG4bK325190B

From: <2132>

namuseum_CUCME#undebug all

All possible debugging has been turned off

namuseum_CUCME#144809@X.X.gr>;tag=10CEAFC4-1376

To: <2132144813>;tag=7945F48C-1F29

Date: Mon, 27 May 2013 13:06:25 GMT

Call-ID: BA311B85-C60411E2-8CD8F11B-22DC6BD2@X.X.gr

Timestamp: 1369659838

CSeq: 101 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

May 27 13:03:58.064: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:

Sent:

ACK sip:2132144813@X.X.16.153:5060 SIP/2.0

Via: SIP/2.0/UDP X.X.71.34:50

namuseum_CUCME#60;branch=z9hG4bK325190B

From: <>2132144809@X.X.gr>;tag=10CEAFC4-1376

To: <2132144813>;tag=7945F48C-1F29

Date: Mon, 27 May 2013 13:03:58 GMT

Call-ID: BA311B85-C60411E2-8CD8F11B-22DC6BD2@X.X.gr

Max-Forwards: 70

CSeq: 101 ACK

Allow-Events: telephone-event

Content-Length: 0

How can the outgoing dial peer work for phones and fail for AA? Do I miss something?Thanks for any help.