Solved: AAR not working " network congestion rerouted" its not rerouting out of pstn

monasir · ‎08-16-2013

Hi

I'm trying to setup AAR fucntionality but its not working

I did the right steps but after testing i can see the call goes out of the gateway but i hear a busy tone.

What i did was the following:

Created aar partitions

Created aar css's

created aar group

created route pattern and route list for aar.

Enabled AAR in service parameters to => true

I'm testing between 2 sites ( registered to UCM ) site a & b

In 2 sites i applied AAR group to the DN's

In site XX i applied AAR to the phone itself

i point the router pattern to the partition which i've created earlier ( pt-aar)

also the route list (rl-aar)

checked ( calling party external phone numbers mask)

transform mask===> XXXXXXXXXX ( 10 dgits)

Predot ( national - isdn)

If i reduce the bandwith in the locations i receive the message on my phone ( Network congestion rerouted)
after that i receive a busy tone so thats means its not rerouting out of the pstn as a long distance call.

I've also resetted the gateways in UCM, resyncd the bandwith no success

Here is my output of the gateways:

SiteA-RTR#

Aug 16 22:02:23.827: ISDN Se0/2/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0002

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98388

Exclusive, Channel 8

Display i = 'SiteA Phone 1'

Calling Party Number i = 0x0081, '2025552001'

Plan:Unknown, Type:Unknown

Called Party Number i = 0xA1, '4083873002'

Plan:ISDN, Type:National

Aug 16 22:02:23.867: ISDN Se0/2/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x8002

Channel ID i = 0xA98388

Exclusive, Channel 8

Aug 16 22:02:23.923: ISDN Se0/2/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x8002

Cause i = 0x8281 - Unallocated/unassigned number

Aug 16 22:02:24.499: ISDN Se0/2/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x0002

Aug 16 22:02:24.507: ISDN Se0/2/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x8002

SiteB-RTR#

Aug 16 22:02:23.863: ISDN Se0/2/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0085

Bearer Capability i = 0x8090A2

Standard = CCITT

Transfer Capability = Speech

Transfer Mode = Circuit

Transfer Rate = 64 kbit/s

Channel ID i = 0xA98381

Exclusive, Channel 1

Display i = 'SiteA Phone 1'

Calling Party Number i = 0x2181, '2025552001'

Plan:ISDN, Type:National

Called Party Number i = 0xA1, '4083873002'

Plan:ISDN, Type:National

Aug 16 22:02:23.863: ISDN Se0/2/0:23 Q931: Received SETUP callref = 0x8085 callID = 0x0006 switch = primary-ni interface = User

Aug 16 22:02:23.879: ISDN Se0/2/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8085

Channel ID i = 0xA98381

Exclusive, Channel 1

Aug 16 22:02:23.883: ISDN Se0/2/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x8085

Cause i = 0x8081 - Unallocated/unassigned number

Aug 16 22:02:23.895: ISDN Se0/2/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x0085

Aug 16 22:02:23.895: ISDN Se0/2/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8085

Nishant Savalia · ‎08-19-2013

Hi,

From debug you can have a clear vision that incoming dial-peer being selected is dial-peer 1 and that dial-peer doesn't do any kind of modification.

Try to implement below solution and test.

voice-port 0/2/0:23

translation-profile incoming PSTN

exit

voice translation-rule 100

rule 1 /.*$3...$/ /\1/

exit

voice translation-profile PSTN

translate called 100

exit

Regards, Nishant Savalia

View solution in original post

Nishant Savalia · ‎08-19-2013

Hi,

Great that it worked.

Actualy the problem is with the lack of configuration.

As your SiteB gateway is H.323 gateway and gateway is sending call to CUCM with help of 2 voip dial-peers with destination-pattern 3... and the call coming from pstn was of 10 digit.

So to match that voip dial-peer you have to convert 10 digit to 4 digit so that required extension can be triggered.

Regards, Nishant Savalia

View solution in original post

Chris Deren · ‎08-17-2013

Are you allowed to dial this 408 number as 10 digits? Did you try as 11 digits? Can you call this number from the phone?

Chris

monasir · ‎08-17-2013

Hi Chris,

I tried it,

Get the same thing busy tone but thats because i don't have an existing routepattern/dial-peer

Chris Deren · ‎08-17-2013

Area code 408 requires 11 digit dialing, see:

http://www.nanpa.com/enas/area_code_query.do

so, make sure your AAR group prefixes 1 (along with off-net access code if dial-peers/route patterns have it).

Chris

monasir · ‎08-17-2013

chris,

i'm sorry but i'm not following you

forgot to mention this is for lab purposes

Chris Deren · ‎08-17-2013

You are sending the call with 10 digits instead of telco expected 11, hence the telco unallocated error, you need to prefix 1 via the AAR group.

Sent from Cisco Technical Support iPhone App

monasir · ‎08-17-2013

Chris,

Thanks for your quick reply..

I wil try this and let you know right nowe not at home.

monasir · ‎08-18-2013

Hi chris,

If i put 1 as prefix in the aar group the call doesnt even to attempt

to come in the gateway.

i"ll send some screenshot of my rp

that points to the gateway

Nishant Savalia · ‎08-18-2013

Hi,

Can you please share what gateway you are using for Site A and Site B?

Because your Site B gateway receives the call from Site A via PSTN and then releases the call as " Unallocated / unassigned number".

Also if possible do share RTMT detailed trace for the same call.

Regards, Nishant Savalia

monasir · ‎08-18-2013

Hi nishant

Site A is an MGCP gateway

Site B is an H323 gateway

the routelist i've created is based on standard local route group.

The routepattern that i've created ( \+1.408387XXXX ) points to this routelist

I'm not using TEHO for this construction

specs RP:

Calling party checked ( XXXXXXXXXX) 10 digits

Called party ( Predot)

With National /ISDN

Enough info?

Chris Deren · ‎08-18-2013

If you are stripping preDot (+1) how are you sending the 1 to telco then? You should not strip pre dot for this call and need to ensure you have proper dial peer on the GW that matches it.

HTH,

Chris

monasir · ‎08-18-2013

still not working

9.1408387XXXX

dial-peer voice 1 pots

destination-pattern 9[1-9].........

port 0/2/0:23

forward-digits all

Nishant Savalia · ‎08-18-2013

Hi,

I think the extension for SiteB Phones will be of 4 digit..right?

Where you are doing digit manipulation? I mean at router level or in CUCM?

Try to run debug voice ccapi ind 2 and debug h225 asn1 in SiteB router so that you can confirm yourself what digits you are sending towards CUCM.

Have you applied any translation-rule in Site B router? Please do share it if any

Regards, Nishant Savalia

monasir · ‎08-19-2013

Hi nishant i have some voice translation rules configured on my siteB router.

I will send you the running config.

output of debug voice ccapi and h225 doesnt give me more info

SiteB-RTR#

Aug 19 11:26:23.336: //-1/03355D9E8004/CCAPI/cc_api_call_setup_ind_common:

Interface=0x68B7244C, Call Info(

Calling Number=2025552001,(Calling Name=)(TON=National, NPI=ISDN, Screening=User, Passed, Presentation=Allowed),

Called Number=4083873002(TON=National, NPI=ISDN),

Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE,

Incoming Dial-peer=1, Progress Indication=NULL(0), Calling IE Present=TRUE,

Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1

SiteA-RTR#sh run

Building configuration...

Current configuration : 3005 bytes

!

! Last configuration change at 06:11:32 EST Mon Aug 19 2013

!

version 12.4

service timestamps debug datetime msec

service timestamps log datetime msec

no service password-encryption

!

hostname SiteA-RTR

!

boot-start-marker

boot-end-marker

!

logging message-counter syslog

!

no aaa new-model

clock timezone EST -5

network-clock-participate wic 2

network-clock-select 1 T1 0/2/0

dot11 syslog

ip source-route

!

ip cef

no ipv6 cef

!

multilink bundle-name authenticated

!

isdn switch-type primary-ni

!

voice-card 0

!

archive

log config

hidekeys

!

controller T1 0/2/0

pri-group timeslots 1-4,24

!

controller T1 0/2/1

channel-group 0 timeslots 1-24

description WAN CONNECTION

!

interface Loopback0

ip address 10.10.110.2 255.255.255.255

ip ospf network point-to-point

h323-gateway voip bind srcaddr 10.10.110.2

!

interface FastEthernet0/0

no ip address

shutdown

duplex auto

speed auto

!

interface FastEthernet0/1

no ip address

shutdown

duplex auto

speed auto

!

interface FastEthernet0/1/0

switchport trunk native vlan 201

switchport mode trunk

switchport voice vlan 202

!

interface FastEthernet0/1/1

switchport trunk native vlan 201

switchport mode trunk

switchport voice vlan 202

!

interface FastEthernet0/1/2

switchport trunk native vlan 201

switchport mode trunk

switchport voice vlan 202

!

interface FastEthernet0/1/3

switchport trunk native vlan 201

switchport mode trunk

switchport voice vlan 202

!

interface Serial0/2/0:23

no ip address

encapsulation hdlc

isdn switch-type primary-ni

isdn incoming-voice voice

isdn map address ^[2-9]......$ plan unknown type unknown

isdn bchan-number-order ascending

isdn outgoing display-ie

no cdp enable

!

interface Serial0/2/1:0

no ip address

encapsulation frame-relay IETF

frame-relay lmi-type ansi

!

interface Serial0/3/0

ip address 10.10.111.2 255.255.255.0

encapsulation ppp

ip ospf mtu-ignore

no fair-queue

clock rate 2000000

!

interface Serial0/3/1

no ip address

shutdown

clock rate 2000000

!

interface Vlan1

no ip address

!

interface Vlan202

ip address 10.10.201.1 255.255.255.0

!

router ospf 1

log-adjacency-changes

network 0.0.0.0 255.255.255.255 area 0

!

ip forward-protocol nd

no ip http server

no ip http secure-server

!

disable-eadi

!

control-plane

!

voice-port 0/2/0:23

!

ccm-manager fax protocol cisco

!

dial-peer voice 1 pots

incoming called-number .

direct-inward-dial

!

dial-peer voice 2 voip

destination-pattern 3...

voice-class codec 1

voice-class h323 1

session target ipv4:10.10.210.11

incoming called-number .

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 3 voip

destination-pattern 3...

voice-class codec 1

voice-class h323 1

session target ipv4:10.10.210.10

dtmf-relay h245-alphanumeric

no vad

!

dial-peer voice 911 pots

destination-pattern 911

port 0/2/0:23

forward-digits 3

!

dial-peer voice 7 pots

translation-profile outgoing 7

destination-pattern 9[2-9]......

port 0/2/0:23

forward-digits 7

!

gatekeeper

shutdown

!

line con 0

exec-timeout 0 0

password cisco

login

line aux 0

line vty 0 4

exec-timeout 0 0

password cisco

login

transport input telnet

!

scheduler allocate 20000 1000

ntp server 10.10.110.1

end

Nishant Savalia · ‎08-19-2013

Hi,

From debug you can have a clear vision that incoming dial-peer being selected is dial-peer 1 and that dial-peer doesn't do any kind of modification.

Try to implement below solution and test.

voice-port 0/2/0:23

translation-profile incoming PSTN

exit

voice translation-rule 100

rule 1 /.*$3...$/ /\1/

exit

voice translation-profile PSTN

translate called 100

exit

Regards, Nishant Savalia