IPIPGW Example

We setup a IPIPGW with 3 gatekeepers and 2 callmanagers.

IPPhone1—Callmanager 1—–GK1———–(IPIPGW/VIAZONEGK)———–GK2———Callmanager2—IPPhone2

GK1 configs:

int lo 0

ip address 172.12.100.1 255.255.255.0 

gatekeeper
 zone local HQ-RTR ipexpert.com 172.12.100.1
 zone remote VIAZONE ipexpert.com 10.12.200.2 1719
 zone prefix VIAZONE 20*
 gw-type-prefix 1#* default-technology
 no shutdown
!

GK2 Configs:

gatekeeper
 zone local LT-RTR ipexpert.com 192.168.10.46
 zone remote VIAZONE ipexpert.com 10.12.200.2 1719
 zone prefix VIAZONE 40*
 gw-type-prefix 1#* default-technology
 no shutdown

IPIPGW config:

 voice service voip
 allow-connections h323 to h323

dial-peer voice 1010 voip
 destination-pattern 20..
 session target ras
 incoming called-number 40..
 dtmf-relay h245-alphanumeric
 codec transparent
!
dial-peer voice 1020 voip
 destination-pattern 40..
 session target ras
 incoming called-number 20..
 dtmf-relay h245-alphanumeric
 codec transparent
!

int fa0/0

ip route-cache same-interface

h323-gateway voip interface
h323-gateway voip id VIAZONE ipaddr 10.12.200.2 1719
h323-gateway voip h323-id PSTNSw
h323-gateway voip tech-prefix 1#

VIAZONE GK Configs:

gatekeeper
 zone local VIAZONE ipexpert.com 10.12.200.2
 zone remote HQ-RTR ipexpert.com 172.12.100.1 1719 invia VIAZONE outvia VIAZONE
 zone remote LT-RTR ipexpert.com 192.168.10.46 1719 invia VIAZONE outvia VIAZON
E
 zone prefix LT-RTR 20*
 zone prefix HQ-RTR 40*
 no shutdown

Issues with gatekeeper zone prefix and end point registration

If you define a zone prefix for a local zone , then it will prevent gateways from registering using E.164 ID to that zone.

zone local zone10 aaa.com 10.1.1.1
zone prefix zone10 5000

With this configuration, none of the gateways can register to zone10 using E.164 ID’s.

Creating a loopback dial-peer for testing

The IPIPGW supports configuration of an RTP loopback dial peer for use in verifying and troubleshooting H.323 networks. When a call encounters an RTP loopback dial peer, the gateway automatically signals call connect and loops all voice data back to the source. In contrast to normal calls through the VoIP-to-VoIP gateway, RTP loopback calls consist of only one call leg.

dial-peer voice 30 voip
destination-pattern 2222
session target loopback:rtp
incoming called-number 2222
codec g711ulaw

Telephony call-legs: 0
SIP call-legs: 0
H323 call-legs: 1
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 1

13FB : 67 97945490ms.1 +20 pid:30 Answer 5004 active
dur 00:00:12 tx:631/100960 rx:631/100960
IP 10.200.200.250:17012 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

How to setup B-ACD so that CME phones and FXO ports can access it?

Usually you call the B-ACD application by matching the incoming dial-peer and calling the application using the service command. You can also set it so that you can call from IP Phones and FXO ports directly to AA application

dial-peer voice 10 voip
destination-pattern 1000 (this is the pilot number defined in AA)
codec g711ulaw
session-target ipv4:10.1.1.1 (use a loopback address of the router so that the dial-peer will always stay up)

dial-peer voice 11 voip
incoming called-number 1000
codec g711ulaw
service aa (this dial-peer will invoke the aa application)

If you monitor calls you can actually see three legs for the calls. See below

Telephony call-legs: 1
SIP call-legs: 0
H323 call-legs: 2
Call agent controlled call-legs: 0
SCCP call-legs: 0
Multicast call-legs: 0
Total call-legs: 3

13DD : 403 633897060ms.1 +1490 pid:20011 Answer 5004 active
dur 00:00:11 tx:0/0 rx:543/86880
Tele 50/0/7 (403) [50/0/7.0] tx:9320/9320/0ms g711ulaw noise:0 acom:0 i/0:0/0 dBm

13DD : 404 633898520ms.1 +30 pid:204 Originate 5060 active
dur 00:00:11 tx:543/86880 rx:316/50560
IP 10.200.200.250:17668 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

13DD : 405 633898530ms.1 +10 pid:205 Answer 5004 active
dur 00:00:11 tx:316/50560 rx:543/86880
IP 10.200.200.250:19166 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a

IPCC High Availability

When you install two IPCC servers the following considerations hold true:

 a. The first server you activate (CRS Engine and/or Datastore services) will become the active CRS box.

 b. The second server you activate (CRS Engine) will become the standby box for the active CRS engine.

 c. The second server you activate (Datastore engine) will become the standby box for the active Datastore engine. (Agent, Historical etc).

 d. When you create a jtapi user in the active box (first server in your cluster), the user name that is created in Callmanager is “jtapi_1″ the one signifying that its the jtapi user for the first CRS server. When you install the second CRS server, it will create another jtapi user named “jtapi_2″.

 e. The second server that you install will use the administrator username / password of the first box during the initial server setup. You should not use Administrator / ciscocisco password.

 f. If the active box has a jtapi group with 20 CTI ports ranging from 3201 to 3220, when you initialize the standby box, it will automatically create 20 more CTI ports from 3221 to 3240. 

 g. JTAPI groups that are created in IPCC coressponds to CTI ports created in Callmanager. The ports in Callmanager that belong to the active CRS box will have a description (if you leave it at defaults) as JTAPI Group #0 – 1. The ports that belong to the standby CRS box will have a description JTAPI Group #0 – 2

CME – using transcoding resources on a separate chassis

You can use transcoding resources available on a separate chassis with CME. You can go upto 5 sdspfarm sessions. CME will share all available resources.

CME configs:

ip source-address 10.1.11.4 port 2000 ———– 10.1.11.4 is the CME router.

sdspfarm units 2
sdspfarm transcode sessions 2
sdspfarm tag 1 mtp0013c3443150 —- this transcoder is on the same box as CME. Lets call this trancoder 1. This is on NM-HDV2
sdspfarm tag 2 mtp0009b7da6540 —- this transcoder is on a different box. Lets call this transcoder 2. This is on NM-HDV

Transcoder 1 configs:

voice-card 1
dspfarm
dsp service dspfarm

sccp local FastEthernet0/0
sccp ccm 10.1.11.4 identifier 1
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register mtp0013c3443150
!
dspfarm profile 1 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec gsmfr
codec g729r8
codec g729br8  ————— g729r8 and g729br8 are not there by default
maximum sessions 1
associate application SCCP

Transcoder 2 configs:

voice-card 1
dsp service dspfarm

sccp local FastEthernet0/0
sccp
sccp ccm 10.10.10.1 priority 1
!
dspfarm transcoder maximum sessions 4
dspfarm
!

Two calls were placed to force the transcoding and CME did utilize each transcoder (The number of transcoding sessions per transcoder was set to 1)

sh sccp connection output on Transcoder 1: (call1)

PSTNSw#sh sccp conn
sess_id conn_id stype mode codec ripaddr rport sport

1 1 xcode sendrecv g729 10.1.11.4 2000 17040
1 2 xcode sendrecv g711u 10.1.11.4 2000 16416

Total number of active session(s) 1, and connection(s) 2
sh sccp connection output on Transcoder 2: (call2)

br2#sh sccp conn
sess_id conn_id stype mode codec ripaddr rport sport

3 3 xcode sendrecv g729 10.1.11.4 2000 18014
3 4 xcode sendrecv g711u 10.1.11.4 2000 19080

Total number of active session(s) 1, and connection(s) 2

After hours block pattern (significance of time)

Lets look at an example where you want to block international calls outside of business hours 8 am to 5 pm.

Remember the time you should configure is in 24 hours format. So 5 pm is 17:00.

after-hours block pattern 1 9011
after-hours day Sun 17:01 08:59
after-hours day Mon 17:00 08:59
after-hours day Tue 17:00 08:59
after-hours day Wed 17:00 08:59
after-hours day Thu 17:00 08:59
after-hours day Fri 17:00 17:00
after-hours day Sat 17:01 17:00

If you specify the time as 08:59, it includes the 60 seconds from 08:59:00 to 08:59:59. So calls will be blocked until 08:59:59. If you specify the time as 17:00, calls will be blocked till until 17:00:59.

CME – Overlay vs Overlay with call waiting

Scenario 1

Two phones need to have a shared line. When a call is answered on the first phone and a second call comes in to the shared line, you need to have the call ring at second phone with no call waiting on the first.

ephone-dn 1
number 3001
huntstop channel
no huntstop

ephone-dn 2
number 3001
huntstop channel
preference 1
ephone 1
button 1o1,2

ephone 2
button 1o1,2

Scenario 2

Two phones need to have a shared line. When a call is answered on the first phone and a second call comes in to the shared line, you need to have the call ring at second phone with call waiting on the first.

ephone-dn 1
number 3001
huntstop channel
no huntstop

ephone-dn 2
number 3001
huntstop channel
preference 1
ephone 1
button 1c1,2

ephone 2
button 1c1,2

How to speed up CTI port addition process on IPCCX?

When you add CTI ports via IPCCX admin, it uses AXL to add into CCM database.

The service parameter “Maximum AXL Writes Allowed per Minute” determines how soon it can add to the database. By default it is 20. Give a higher value to speed up this process.

IPCCX – provisioning CTI port and Media termination groups

CTI ports take only 1 ms to reinitialize (ie, to get back to operation after a call is disconnected), but CMT groups will take upto 200 ms to reinitialize. IPCCX will not answer calls if there are no CMT ports available. So make sure you provision 10% more CMT ports than CTI ports.

VT advantage – unlimited

VTA uses CAST protocol. Uses 4224 TCP port. This protocol helps in discovering remote VTA capable endpoints, communicates with IP phone as well as with Callmanager. To ensure proper QOS is given to CAST, on the phone port, create a service policy to mark TCP 4224 with DSCP CS3.

VTA uses two types of codecs.

a. H263 – min of 128kbps to a max of 1.5 Mbps per call

b. Wideband – 7.0 mbps per call

Callmanager – AD integration – Adding /Deleting users from CCM admin page

Step 1 Browse to C:\dcdsrvr\config, and open the file, UMDirectoryConfiguration.ini, in Notepad.

---

Caution You must open the file in Notepad. Opening the file in another text editor application may corrupt the file.

---

Step 2 In the UMDirectoryConfiguration.ini file, locate the key, UserDirAccess.

Step 3 Change the UserDirectAccess value to true.

Step 4 In Notepad, choose File > Save to save the file.

Step 5 Close the UMDirectoryConfiguration.ini file.

Step 6 Restart the IIS Admin Service and its dependent services by choosing Start > Programs > Administrative Tools> Services.

Step 7 Right-click IIS Admin Service and then choose Restart.

Step 8 A dialog box prompts you to restart dependent services. These services may differ depending on your configuration. Click Yes.

Step 9 Restart the dependent services.

QoS for CUE CCM Integration

One of the current limitations is that the JTAPI (CTI-quick buffer encoding [QBE]) signaling packets are unmarked (TOS = 0) when the Cisco Unity Express AIM transmits them. In order to correct this, use an access control list (ACL) on the router that has the Cisco Unity Express AIM installed to mark and prioritize the traffic.

The JTAPI signaling from the Cisco CallManager is correctly marked with a differentiated services code point (DSCP) value of CS3.

The JTAPI signaling protocol uses TCP port 2748. Dedicate 20 kbps per Cisco Unity Express site for this traffic.

All Real-Time Protocol (RTP) audio traffic from the Cisco Unity Express AIM or the IP phone is correctly marked with a DSCP value of 0xEF.

This example shows a sample configuration for this on the router where a.b.c.d is the IP address of the Cisco Unity Express AIM:

access-list 101 permit tcp host a.b.c.d any eq 2748
!
class-map match-all cti-qbe
 match access-group 101
!
policy-map cti-qbe
 class cti-qbe
 set dscp af31
 bandwidth 20
!
interface Serial0/1
 service-policy output cti-qbe

Why CUE integration with CCM doesn’t need MWI numbers?

Since Cisco Unity Express uses JTAPI to interface with the Cisco CallManager, no MWI on and off numbers need be configured.

Callforward behaviour in 4.1

Callforward All

If the Call Forward all CSS is set to none, it will concatenate the line and device CSS. This applies to settings done via the phone as well as via the CCM user page.

Callforward Busy/No-answer 

If the Callforward CSS is set to none, it does not concatenate the line and device CSS.

How to prepend digits to called id in MGCP

Based on the ISDN number plan and type of the incoming call, the following values will be used to prepend the Caller ID.

National number prefix

International number prefix

Subscriber number prefix

Unknown number prefix

Overlap receiving flag for PRI

To enable Overlap receiving for PRI interfaces that you the PRI Backhaul mechanism

Matching calling party with Attendant DN

If an Attendant DN is configured on FXO port and this parameter is set to True, then Callmanager will limit outbound calls from that DN to only use the FXO port that is configured with that Attendant DN. You would need to create a route pattern and include this FXO port in the route list. Even though there are other gateways that could be listed above in the route list, it would still use this gateway only.

Also, none of the other phones will be able to use this gateway even though this gateway appears in the route list.

Disable resume from MGCP shared line FXS port

This parameter determines if you can resume a call from hold on a MGCP FXS shared line.

Default value is true

Caller ID

This is a global setting for Caller ID for all gateways. You can also set the Caller ID individually at the gateway level

Line Device CSS contatenation

In CCM 4.1, Callmanager concatenates the CSS of the device and lines and places the CSS of the lines in front of the CSS of the device.

As per SRND, For CTI ports and CTI Route points, it does the other way. ie, CSS of the device comes first followed by the CSS of the lines. This used to be the behaviour in CCM  3.1. This was not the case during testing. Even the CTI ports and CTI route points were using line CSS First

CMC and FAC

When you have a CMC and FAC enabled on a route pattern, you have to enter FAC first, before entering CMC code.

CMC codes do show up on CDR records (unlike FAC)

FAC and CDR

When you define FAC code names and auth codes, only FAC code names appear on CDR. (For added security). The auth codes themselves never show up.

Digit manipulation – best practices

When doing digit manipulation, first decide whether you want to do it at route pattern level or route group level. Route group level is recommended for following reasons.

 a. The end user will see the manipulated digits if you do digit manipulation at the route-pattern level. If you do it at route group level, the end user will not see that on his phone.

b. IF you do it at route pattern level, CDR will log the digits including the manipulated ones. So its recommended to do it at the route group level.

c. Any digit manipulation you do at route group level completely overrides the setting at route pattern level.

MGCP Call Preservation

MGCP PRI backhaul does not support call preservation when transitioned from Callmanager to SRST and vice versa.

H323/MGCP – Caller ID Display

H323 does not support Facility IE. H323 supports only Display IE

MGCP supports both Display and Facility IE

Emergency Alternate Greeting – EAG

EAG is implemented as a system script called checkaltgreet.aef. To activate EAG is a script, insert a Call Subflow checkaltgreet.aef into the script.

EAG is activated by the existence or absence of the greeting. It is stored as AltGreeting.wav. If this file exists on the system, the EAG is automatically activated and the special greeting is played whenever the script directs it to be played.

The number to call to record the greeting (to turn on the EAG feature) or to delete it is the Cisco UE pilot number associated with greeting management system (GMS).

B-ACD Queue-Debugs

Give param queue-manager-debugs 1 and from privilege mode, “debug voip application script”.

 Site2#
*Jul 17 00:44:53.311: //89//TCL :/tcl_PutsObjCmd:
proc init_perCallvars
*Jul 17 00:44:53.311:
*Jul 17 00:44:53.323: //89//TCL :/tcl_PutsObjCmd: TCL AA: ++ Playing Welcome Prompt and options menu ++
*Jul 17 00:44:58.383: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ No option selected +++
*Jul 17 00:44:58.455: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> B-ACD Service Running <<<
*Jul 17 00:45:06.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: +++ New incoming call to queue 3210 +++
*Jul 17 00:45:06.703: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_NEW} {1153097106}} <<<
*Jul 17 00:45:06.703: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: +++ 3210 Queue Length = 1 +++
*Jul 17 00:45:06.719: //89//TCL :/tcl_PutsObjCmd: TCL AA: >>> Destination Huntgroup = 3210 <<<
*Jul 17 00:45:06.747: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_PROGRESS} {1153097106}} <<<
*Jul 17 00:45:06.767: //89//TCL :/tcl_PutsObjCmd: TCL AA: — MEDIA STATUS = ms_001 –
*Jul 17 00:45:06.767: //89//TCL :/tcl_PutsObjCmd: TCL AA: — ERROR: Unable to play prompt; File not found –
*Jul 17 00:45:06.771: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Playing MOH +++
*Jul 17 00:45:16.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_PROGRESS} {1153097116}} <<<
*Jul 17 00:45:16.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:26.699: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_PROGRESS} {1153097126}} <<<
*Jul 17 00:45:26.703: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:30.815: //89//TCL :/tcl_PutsObjCmd:
 proc act_CallSetupDone
*Jul 17 00:45:30.819:
*Jul 17 00:45:30.819: //89//TCL :/tcl_PutsObjCmd:
 IP IVR Leg Setup Status = ls_002
*Jul 17 00:45:30.819:
*Jul 17 00:45:30.823: //89//TCL :/tcl_PutsObjCmd: TCL AA: >> Agent did not answer; Received Call NOAN <<
*Jul 17 00:45:30.835: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_NOAN} {1153097130}} <<<
*Jul 17 00:45:36.695: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Now playing second greeting prompt +++
*Jul 17 00:45:36.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_NOAN} {1153097136}} <<<
*Jul 17 00:45:36.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:45.831: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY} {1153097145}} <<<
*Jul 17 00:45:45.879: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY_PROGRESS} {1153097145}} <<<
*Jul 17 00:45:46.163: //89//TCL :/tcl_PutsObjCmd: TCL AA: — MEDIA STATUS = ms_000 –
*Jul 17 00:45:46.167: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Playing MOH Livefeed +++
*Jul 17 00:45:46.219: //89//TCL :/tcl_PutsObjCmd: TCL AA: — MEDIA STATUS = ms_001 –
*Jul 17 00:45:46.219: //89//TCL :/tcl_PutsObjCmd: TCL AA: — ERROR: Unable to play prompt; File not found –
*Jul 17 00:45:46.223: //89//TCL :/tcl_PutsObjCmd: TCL AA: +++ Playing MOH +++
*Jul 17 00:45:46.711: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY_PROGRESS} {1153097146}} <<<
*Jul 17 00:45:46.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:56.711: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> THE QUEUE IS {TclModule_2227448556_0_5762376 {CALL_RETRY_PROGRESS} {1153097156}} <<<
*Jul 17 00:45:56.715: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Received HELLO from TclModule_2227448556_0_5762376 <<<
*Jul 17 00:45:57.415: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> Call Abandoned Time = 64 Seconds<<<
*Jul 17 00:45:57.427: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD ++ Service Counter = 6 ++
*Jul 17 00:45:57.431: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: >>> Queue 3210 Statistics = {“Total calls presented to the queue: ” 5}
        {“Calls answered by agents: ” 1}
        {“Number of calls in the queue: ” 1}
        {“Average time to answer (secs): ” 20}
        {“Longest time to answer (secs): ” 20}
        {“Number of abandoned calls: ” 4}
        {“Average time before abandon (secs): ” 133}
        {“Calls forwarded to voice mail: ” 0}
        {“Calls answered by voice mail: ” 0} <<<
*Jul 17 00:45:57.435: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD: +++ 3210 Queue Length = 0 +++
*Jul 17 00:45:57.439: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> 3210 Queue Length = 0 <<<
*Jul 17 00:45:57.439: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> Queue 3210 is Empty <<<
*Jul 17 00:45:58.455: //40//TCL :/tcl_PutsObjCmd: TCL B-ACD:  >>> B-ACD Service Running <<<

Keep Conference in CME

By default if conference initiator hangs up the call, the conference is terminated.

To keep the conference parties connected when the conference initator leaves, configure under the  ephone “keep-conference“. When the initiator hangsup the remaining parties stay connected. If he uses the end-call soft key, the conference is terminated.

To keep the conference parties connected when the conference initator leaves by hanging up or pressing the end-call softkey, use “keep- conference endcall” option under ephone

CME – Caller ID block

For both internal and external calls

ephone-dn

caller-id block

* This does not apply for E1R2, because caller id blocking is a flag in the setup messages and E1R2 does not support that. This does not apply for FXO, because caller id is controllerd by Telco on FXO. Use translation patterns to change caller ID to null

Only external calls

dial-peer voice 10 pots

clid strip – to block only the number

clid strip name – to block only the name

To send only caller name

telephony-service

caller-id name-only

To block caller id on a call by call basis

telephony-service

caller-id block code *67