09-25-2013 08:56 AM - edited 03-16-2019 07:32 PM
On my lab, from a CorpHQ phone, when I test dial-peers for outbound dialing, only 911 & 9911 work. When I go directly to the Gateway and type the CSIM START command, the 911 pattern works and the 912065015111 NATIONAL pattern works, however the 92065015111 LOCAL pattern geta an UNASSIGNED/UNALLOCATED NUMBER error even though Dial-peer 11 is being matched. Please see the configs and debug.
Thanks,
Ron
___________________________________________________________________________________________________________________
dial-peer voice 10 pots
destination-pattern 911
no digit-strip
port 0/0/0:23
!
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
_________________________________________________________________________________________________
CorpHQ#
//-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Calling Number=, Called Number=92065015111, Peer Info Type=DIALPEER_INFO_SPEECH
//-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Match Rule=DP_MATCH_DEST; Called Number=92065015111
//-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Result=Success(0) after DP_MATCH_DEST
//-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
Result=SUCCESS(0)
List of Matched Outgoing Dial-peer(s):
1: Dial-peer Tag=11
ISDN Se0/0/0:23 Q931: pak_private_number: Invalid type/plan 0x0 0x0 may be overriden; sw-type 13
ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x00A6 callID = 0x8027 switch = primary-ni interface = User
CorpHQ#
ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x00A6
Bearer Capability i = 0x8090A2
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA98383
Exclusive, Channel 3
Called Party Number i = 0xA1, '2065015111'
Plan:ISDN, Type:National
ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x80A6
Channel ID i = 0xA98383
Exclusive, Channel 3
ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x80A6
Cause i = 0x8281 - Unallocated/unassigned number
ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x00A6
ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80A6
Solved! Go to Solution.
09-26-2013 01:20 PM
As stated by you, 11 pots is not working and 12 pots is working
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
now when i see debugs , i see that telco is sending the disconnect when your 11 pots dial-peer is hitting
in the debugs i see that , called party number is 10 digit, type :Nat and Plan:ISDN
the only diffrence that i see that is this
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
the reason above is working because your telco expects a leading digit for a 10 digit number
so when 12 pots is selected , telco is reciving "1206XXXXXXX" and telco is not reciving this in case of 11 pots
hence it is sending a disconnect
to fix this issue , simply do a prefix 1 on 11 pots and there is no way that call will not work , given the fact that 12 pots is working correctly
regards
Anurag Siddhu
09-25-2013 09:05 AM
Hi Ron,
You are receiving disconnect message from peer side, because you are sending called number type as "National". It should be local as per your requirement.
Please check your configuration.
However, how do you use the same number for National & Local for the same calling number?
Regards,
Nishant Savalia
09-25-2013 11:35 AM
Nishant:
Do you mean create a voice translation rule to change the called number type?
Thanks,
Ron
09-25-2013 11:02 PM
Hi Ron,
Yes, I mean to create a voice translation profile at gateway level and apply it under the selected dial-peer.
If you don't apply any translation rule and if you are sending 10 digit as called number then IOS by default will send that number as National number. This thing happening with your current configuration.
And as you are trying for local number, it will be 7 or 8 digits. Local doesn't need area code (for e.g. 206)
What i would suggest is to check the requirement of peer or remote side because call is being released from remote end.
Also, please check the called number as you are using the same called number for National & Local as well (2065015111).
National number will be different from the local number. Please check this first.
Regards,
Nishant Savalia
09-25-2013 11:12 PM
Hello Nishant and RON
You are completely true for Area code , and the local number (7 or 8) . Some labs for rack rental make the local calls 10 digits , as example INE rack rental the local call is 10 digits , so this is a simple lab .Please RON share your configuration , your route pattern for local call.
Thank you
please rate if this will help
09-26-2013 06:33 AM
CorpHQ#sh run
Building configuration...
Current configuration : 5608 bytes
!
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
!
hostname CorpHQ
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
clock timezone PST -8
clock summer-time PDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
!
dot11 syslog
ip source-route
!
!
ip cef
ip dhcp excluded-address 177.1.11.1 177.1.11.14
ip dhcp excluded-address 177.1.11.21 177.1.11.254
ip dhcp excluded-address 177.2.11.1 177.2.11.14
ip dhcp excluded-address 177.2.11.21 177.2.11.254
!
ip dhcp pool CorpHQ-Phones
network 177.1.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.1.11.1
dns-server 177.1.100.110
!
ip dhcp pool Branch1-Phones
network 177.2.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.2.11.1
dns-server 177.1.100.110
!
!
no ip domain lookup
ip multicast-routing
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
!
voice service voip
allow-connections h323 to h323
fax protocol cisco
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 10
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011/ /9&/
!
!
voice translation-profile Prefix9_InFrom_CUCM
translate called 10
!
!
voice-card 0
dsp services dspfarm
!
!
!
!
!
archive
log config
hidekeys
!
!
!
!
!
controller T1 0/0/0
pri-group timeslots 1-3,24
description == Voice Circuit to PSTN
!
!
!
!
!
interface Loopback0
ip address 177.1.254.1 255.255.255.255
ip pim dense-mode
!
interface FastEthernet0/0
description == To CorpHQ-Switch
no ip address
duplex auto
speed auto
!
interface FastEthernet0/0.10
description == Server VLAN
encapsulation dot1Q 10
ip address 177.1.10.1 255.255.255.0
ip pim dense-mode
!
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.1.11.1 255.255.255.0
ip helper-address 177.1.10.10
ip nbar protocol-discovery
ip pim dense-mode
!
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.1.12.1 255.255.255.0
!
interface FastEthernet0/0.13
description == PSTN PHONE VLAN
encapsulation dot1Q 13
ip address 177.1.13.1 255.255.255.0
!
interface FastEthernet0/1
description === To PSTN
ip address 177.1.19.254 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 36
cdp enable
frame-relay lmi-type ansi
ip rsvp bandwidth
!
interface Serial0/1/0.1 point-to-point
description == FR To BR1
bandwidth 384
ip address 177.0.101.1 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
!
interface Serial0/1/0.2 point-to-point
description == FR To BR2
ip address 177.0.201.1 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
!
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 177.1.19.1
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0.10
no ip http server
no ip http secure-server
!
!
!
!
!
!
!
!
!
control-plane
!
!
!
voice-port 0/0/0:23
!
voice-port 0/3/0
!
voice-port 0/3/1
!
ccm-manager music-on-hold
!
!
sccp local Loopback0
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp
!
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 3 register CorpHQ-HW-Xcode
associate profile 2 register CorpHQ-711-MTP
associate profile 1 register CorpHQ-729-MTP
!
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 2
associate application SCCP
!
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
!
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
!
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 10 pots
destination-pattern 911
no digit-strip
port 0/0/0:23
!
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
!
dial-peer voice 100 voip
description == Inbound/Outbound SIP PSTN GW From/To CUCM Pub
translation-profile incoming Prefix9_InFrom_CUCM
destination-pattern ^2065011...$
voice-class codec 1
session protocol sipv2
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
!
!
dial-peer hunt 1
sip-ua
!
!
!
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
!
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 2
ntp server 177.1.254.254
end
09-25-2013 09:24 AM
Hello
Can you use the below configuration and try again.
voice translation-rule 2
rule 1 // // type any subscriber plan any isdn
voice translation-profile local
translate called 2
dial-peer voice 11 pots
translation-profile outgoing local
destination-pattern 9[2-9]..[2-9]......$
forward-digits 10
port 0/0/0:23
if this not work , please share with us your all configuration. Tell us also please this configuration for CME , or CUCM and H323 GW .
Thank you
please rate all useful information
09-25-2013 10:33 AM
Hi Islam:
The voice translation rule didn't work. Please see the following...
Thanks,
09-25-2013 10:48 AM
Did you verify that the number is dialable as 10 digits and not 7 digits or 11 digits for example?
Chris
09-25-2013 07:50 PM
Hi Ron.
Can you please post you actual running config?
Thanks
Regards
Carlo
Sent from Cisco Technical Support iPhone App
09-26-2013 06:29 AM
CorpHQ#sh run
Building configuration...
Current configuration : 5608 bytes
!
version 12.4
no service pad
no service timestamps debug uptime
no service timestamps log uptime
no service password-encryption
!
hostname CorpHQ
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
clock timezone PST -8
clock summer-time PDT recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
!
dot11 syslog
ip source-route
!
!
ip cef
ip dhcp excluded-address 177.1.11.1 177.1.11.14
ip dhcp excluded-address 177.1.11.21 177.1.11.254
ip dhcp excluded-address 177.2.11.1 177.2.11.14
ip dhcp excluded-address 177.2.11.21 177.2.11.254
!
ip dhcp pool CorpHQ-Phones
network 177.1.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.1.11.1
dns-server 177.1.100.110
!
ip dhcp pool Branch1-Phones
network 177.2.11.0 255.255.255.0
option 150 ip 177.1.10.10 177.1.10.20
default-router 177.2.11.1
dns-server 177.1.100.110
!
!
no ip domain lookup
ip multicast-routing
no ipv6 cef
!
multilink bundle-name authenticated
!
!
!
!
isdn switch-type primary-ni
!
!
!
voice service voip
allow-connections h323 to h323
fax protocol cisco
sip
bind control source-interface Loopback0
bind media source-interface Loopback0
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!
!
!
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 10
rule 1 /^[2-9].........$/ /9&/
rule 2 /^1[2-9].........$/ /9&/
rule 3 /^011/ /9&/
!
!
voice translation-profile Prefix9_InFrom_CUCM
--More--
%PIM-5-NBRCHG: neighbor 177.0.101.2 UP on interface Serial0/1/0.1
%OSPF-5-ADJCHG: Process 1, Nbr 177.1.254.3 on Serial0/1/0.2 from LOADING to FULL, Loading Done
%OSPF-5-ADJCHG: Process 1, Nbr 177.1.254.2 on Serial0/1/0.1 from LOADING to FULL, Loading Done
translate called 10
!
!
voice-card 0
dsp services dspfarm
!
!
!
!
!
archive
log config
hidekeys
!
!
!
!
!
controller T1 0/0/0
pri-group timeslots 1-3,24
description == Voice Circuit to PSTN
!
!
!
!
!
interface Loopback0
ip address 177.1.254.1 255.255.255.255
ip pim dense-mode
!
interface FastEthernet0/0
description == To CorpHQ-Switch
no ip address
duplex auto
speed auto
!
interface FastEthernet0/0.10
description == Server VLAN
encapsulation dot1Q 10
ip address 177.1.10.1 255.255.255.0
ip pim dense-mode
!
interface FastEthernet0/0.11
description == Voice VLAN
encapsulation dot1Q 11
ip address 177.1.11.1 255.255.255.0
ip helper-address 177.1.10.10
ip nbar protocol-discovery
ip pim dense-mode
!
interface FastEthernet0/0.12
description == Data VLAN
encapsulation dot1Q 12
ip address 177.1.12.1 255.255.255.0
!
interface FastEthernet0/0.13
description == PSTN PHONE VLAN
encapsulation dot1Q 13
ip address 177.1.13.1 255.255.255.0
!
interface FastEthernet0/1
description === To PSTN
ip address 177.1.19.254 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
no cdp enable
!
interface Serial0/1/0
description == Frame-Relay Circuit to WAN
no ip address
encapsulation frame-relay
fair-queue 64 256 36
cdp enable
frame-relay lmi-type ansi
ip rsvp bandwidth
!
interface Serial0/1/0.1 point-to-point
description == FR To BR1
bandwidth 384
ip address 177.0.101.1 255.255.255.0
ip pim dense-mode
snmp trap link-status
frame-relay interface-dlci 101
ip rsvp bandwidth 136
!
interface Serial0/1/0.2 point-to-point
description == FR To BR2
ip address 177.0.201.1 255.255.255.0
snmp trap link-status
frame-relay interface-dlci 102
ip rsvp bandwidth 136
!
router ospf 1
log-adjacency-changes
network 0.0.0.0 255.255.255.255 area 0
!
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 177.1.19.1
ip route 0.0.0.0 0.0.0.0 FastEthernet0/0.10
no ip http server
no ip http secure-server
!
!
!
!
!
!
!
!
!
control-plane
!
!
!
voice-port 0/0/0:23
!
voice-port 0/3/0
!
voice-port 0/3/1
!
ccm-manager music-on-hold
!
!
sccp local Loopback0
sccp ccm 177.1.254.3 identifier 3 priority 3 version 5.0.1
sccp ccm 177.1.10.20 identifier 2 priority 2 version 5.0.1
sccp ccm 177.1.10.10 identifier 1 priority 1 version 5.0.1
sccp
!
sccp ccm group 1
bind interface Loopback0
associate ccm 2 priority 1
associate ccm 1 priority 2
associate ccm 3 priority 3
associate profile 3 register CorpHQ-HW-Xcode
associate profile 2 register CorpHQ-711-MTP
associate profile 1 register CorpHQ-729-MTP
!
dspfarm profile 3 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
codec ilbc
maximum sessions 2
associate application SCCP
!
dspfarm profile 1 mtp
codec g729ar8
codec g729r8
rsvp
maximum sessions software 10
associate application SCCP
!
dspfarm profile 2 mtp
codec g711ulaw
rsvp
maximum sessions software 10
associate application SCCP
!
!
dial-peer voice 1 pots
incoming called-number .
direct-inward-dial
!
dial-peer voice 10 pots
destination-pattern 911
no digit-strip
port 0/0/0:23
!
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
!
dial-peer voice 100 voip
description == Inbound/Outbound SIP PSTN GW From/To CUCM Pub
translation-profile incoming Prefix9_InFrom_CUCM
destination-pattern ^2065011...$
voice-class codec 1
session protocol sipv2
session target ipv4:177.1.10.10
incoming called-number .
ip qos dscp cs3 signaling
!
!
dial-peer hunt 1
sip-ua
!
!
!
line con 0
exec-timeout 0 0
privilege level 15
logging synchronous level 0 limit 20
line aux 0
line vty 0 4
exec-timeout 0 0
privilege level 15
logging synchronous
no login
line vty 5 15
exec-timeout 0 0
privilege level 15
logging synchronous
no login
!
scheduler allocate 20000 1000
ntp source Loopback0
ntp master 2
ntp server 177.1.254.254
end
09-26-2013 01:20 PM
As stated by you, 11 pots is not working and 12 pots is working
dial-peer voice 11 pots
destination-pattern 9[2-9]..[2-9]......$
port 0/0/0:23
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
now when i see debugs , i see that telco is sending the disconnect when your 11 pots dial-peer is hitting
in the debugs i see that , called party number is 10 digit, type :Nat and Plan:ISDN
the only diffrence that i see that is this
!
dial-peer voice 12 pots
destination-pattern 91[2-9]..[2-9]......$
port 0/0/0:23
forward-digits 11
the reason above is working because your telco expects a leading digit for a 10 digit number
so when 12 pots is selected , telco is reciving "1206XXXXXXX" and telco is not reciving this in case of 11 pots
hence it is sending a disconnect
to fix this issue , simply do a prefix 1 on 11 pots and there is no way that call will not work , given the fact that 12 pots is working correctly
regards
Anurag Siddhu
09-26-2013 01:59 PM
Anurag:
Excellent job!. It worked, however could this have been corrected in the PSTN configuration instead? I don't want to encounter this issue again...
Many thanks,
Ron
09-26-2013 02:14 PM
yes, if you are simulating this in lab scenario , you can change this config on Telco router and this can be fixed.
if this is real time scenario, you have to contact your telco provider in order to change this.
09-26-2013 02:24 PM
It is a lab, however I am just a novice with voice translation-rules. I wouldn't have a clue what to change on the PSTN router.
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