CME 4FXO Outbound Help

Hello all, I have CME 4.2 and I am trying to get outbound calls to work on my system and I am failing.
I want users to press 9 and then dial their number. However when they hit new call, and then dial 9 they get an immediate fast busy.
I have secondary-dial tone configured for 9
Attached are my dial-peers.

I have
corlist incoming International
corlist outgoing International
on all dial-peers. It is wierd because it immediatly goes to fast busy. The my outgoing dial-peers seem to be working as I can call-forward all to an external number with no problems. I have removed secondary-dial 9 on telephone-service and readd-ed both times re-generating configs.

Similar Messages

CME - Sending outbound calls to FXO port

Hi Guys,
Need your help for the below scenario.
Our customer has a CME where 4 FXO ports are already connected and working. Customer has added 2 more FXO port and few IP phones.
The requirement is when ever an outbound call is made from the newly configured IP phones, the call should go through the newly added FXO lines.
For eg ext 3001 , the outbound call should go through port 0/1/0
Already the prefix 9 is used for dialing the number and I guess only one prefix number can be used in CME.
I tried translation rule , cor list but none worked , the call is default going through the old fxo port and not to the new fxo port.
Can you guys help me with the configuration.
Regards
Sathya

Previous post on similar issue might be helpful -
https://supportforums.cisco.com/discussion/11431746/h323-choose-outbound-fxo-port-based-calling-number
Thnx

Branch Office CME design Verification

Hi All,
Please refer to the attached network diagram.
I need to verify this can be implemented and would work.
We have a branch office moving to a new location and they intend to keep their existing CME (for business reasons), provided by their local service provider with ISDN line for calls to the PSTN. This is managed by the service provider and we have no access to it. However we would like to grant them connectivity to the existing corporate voice network via an IP VPN connection, which shall be put in place soon. This will enable the branch make site to site calls within the corporate network
With a SIP trunk between the internal and external CME, I intend to make all the phones register with the Call Manager, however on the call manager , set a route pattern for calls going out to the PSTN from this branch back to the internal CME and this will then be matched by a SIP dial peer directing the call to the external CME out to the PSTN.
My worry is with the delay that might be introduced when making a PSTN call as the internal CME has to first contact the call manager in order to know where to send the call.
So my questions are as follows,
1. Is this solution feasible especially in terms of delay? If not,
2. Are there any other ways to achieve the same scenario
Thanks,
Yomi

Are the phones at the branch office going to register to the Internal CME? If so, all configuration for outbound dialing will be done on the Internal CME, not on UCM. ie. dial-peer on the Internal CME for outbound dialing. For phone connectivity back to UCM, you will have a SIP trunk between UCM and internal CME and that is perfectly acceptable. You "might" see some quality degradation but that is to be expected from Internet based WAN connectivity. If your RTT delay is greater than 150ms, then you might see some quality issues.

Is it possible to have multiple lines on a phone but only have one of the number ring on that phone?

I have a client who has four number on his phone. Extension 4709 is the clients number. He also has 4715,4725 and 4712 on his phone. Is it possible to make it so only 4709 will ring on his phone? The client is running CME 10.0
New here and new CME so any help would be greatly appreciated

Hi Joseph,
This would be configured using the "button" (s=silent separator) command;
From;
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/command/reference/cme_cr/cme_b1ht.html#wp1050318
Cheers!
Rob
button
To associate ephone-dns with individual buttons on a Cisco Unified IP phone and to specify line type or ring behavior, use the button command in ephone configuration mode. To remove an ephone-dn association from a button, use the no form of this command.
button button-number {separator} dn-tag [,dn-tag...] [button-number{x}overlay-button-number] [button-number...]
no button button-number {separator} dn-tag [,dn-tag...] [button-number{x}overlay-button-number] [button-number...]
Syntax Description
button-number
Number of a line button on a Cisco Unified IP phone that is to be associated with an extension (ephone-dn). The maximum number of button-ephone-dn pairs is determined by the phone type. Note The Cisco Unified IP Phone 7910G has only one physical line button, but you can assign it two button-ephone-dn pairs.
separator
Single character that denotes the characteristics to be associated with this phone button. Valid entries are as follows: •: (colon)—Normal ring. For incoming calls on this extension, the phone produces audible ringing, a flashing icon in the phone display, and a flashing red light on the handset. On the Cisco IP Phone 7914 Expansion Module, a flashing yellow light also accompanies incoming calls. •b—Beep but no ring. Audible ring is suppressed for incoming calls, but call-waiting beeps are allowed. Visible cues are the same as those described for a normal ring. •c—Call waiting. Provides call waiting for secondary calls to an overlaid ephone-dn. See also the o keyword. •f—Feature ring. Differentiates incoming calls on a special line from incoming calls on other lines on the phone. The feature-ring cadence is a triple pulse, as opposed to a single pulse for normal internal calls and a double pulse for normal external calls.
•m—Monitor mode for a shared line. Visible line status indicates whether the line is in-use or not. Monitored lines cannot be used on this phone for incoming or outgoing calls. •o—Overlay line. Multiple ephone-dns share a single button, up to a maximum of 25 on a button. See also the c keyword. •s—Silent ring. Audible ring and call-waiting beep are suppressed for incoming calls. The only visible cue is a flashing ((< icon in the phone display. Note In Cisco IOS Release 12.4(4)XC and later releases, the silent ringing behavior is overridden during active night-service periods. Silent ringing does not apply during designated night-service periods when the s keyword is used. •w—Watch mode for all lines on the phone for which this directory number is the primary line. Visible line status indicates whether watched phone is idle or not.
dn-tag
Ephone-dn tag that was previously defined using the ephone-dn command. When used with the c and o keywords, the dn-tag argument can contain up to 25 individual dn-tags, separated by commas.
x
Separator that creates an overlay rollover button. When the overlay button specified in this command is occupied by an active call, a second call to one of its ephone-dns will appear on this button. This button is also known as an overlay expansion button.
overlay-button-number
Number of the overlay button that should overflow to this button.

How to integrate Oracle Tutor Desk manual Index with Oracle ApplicationsR12

Hi,
I have created a desk manual Index and related HTML files in oracle Tutor 14. Can anyone help me in integrating the desk manual index with the Oracle Applications Help.
My questions are:
-> In what top shd i upload my files using help Upload ?
-> How do i integrate the desk manual index HTML file with my applications help?
Has anyone done this before . If so , Can u please guide me through the process.
Thanks.
Edited by: Vyas on Apr 8, 2012 12:52 AM

Here are some instructions - please let me know if this is what you are looking for.
Kind Regards,
Emily
-> In what top shd i upload my files using help Upload ?
Create Upload and Download Directories on Middle Tier.
     The download path must be a location on the machine that hosts the Application Server - ie, the "middle tier". Directories for the upload (for example, /u01/tutor/upload) and download (for example, /u01/tutor/download) functions must be created on the middle tier.
     ***These directories, as well as any future subdirectories, must have full read/write/execute permissions. The Help Utility will automatically create subdirectories, so the permissions must be set correctly before the Help Utility is used.
     Verify that the HELP_UTIL_DOWNLOAD_DIR directory points to a directory that can be accessed from both the JServ tier and the Concurrent Manager tier.
From the EBS
EBS R12 System Administrator > Profile > System > Find System Profile Values
- Verify Site is checked
- Verify Profiles with No Values is checked
- In the profile field, enter %help%
- Click Find button
Update system profile values with the correct paths.For example:
     Help Utility Download Path > /u01/tutor/download OR /dbfiles/applcsf/outbound
     Help Utility Upload Path > /u01/tutor/upload OR /dbfiles/applcsf/inbound
     Help System Root > FND:LIBRARY
     TCF:Host > http://<hostname.domainname>
     TCF:Port > <port number>
-> How do i integrate the desk manual index HTML file with my applications help?
Use Help Builder to add the DESK_MANUAL_INDEX to the Help Tree
To learn all the features about customizing Help Navigation Trees, refer to the Oracle Applications System Administration User Guide.
EBS R12 System Administration > Help Administration > Help Builder
When Help Builder first opens it may appear as a tiny screen in the upper left corner, if this happens, maximize the screen. You may also have to resize the screen as you may see the entire screen at first.
- Enter FND in the Node Application field.
- Press the Find button on the Find Trees Window.
This should bring the Trees tab region forward with a list of available trees.
- Scroll down and find FND | US | Applications Help Library in the Trees tab
- Select and double click on it.
This will open the Applications Help Library tree in the left frame
- Expand the Application Help Library by clicking on the + next to it.
- Click the New Node icon on the toolbar.
- Enter the following information in Properties window.
The next items assume the default Desk Manual Index terminology is used.
If your organization uses or plans to use different terms, adjust the entries in the various properties field accordingly.
     Prompt: Desk Manual Index
     Description: Desk Manual Index
     Data: @DESK_MANUAL_INDEX
Or enter the exact Desk Manual Index file name if it is different than the example.
- Click the View button to verify the link works.
You may have to experiment with using the @ sign. It may not be required.
     Click the Apply button.
The Desk Manual Index will now appear as a Node under the Application Help Library. If the name is NOT fully displayed, click on the - (minus) sign to expand it.
Link does not work
- Change the information in the Data field.
- Delete the @ and add .htm extension
Example: DESK_MANUAL_INDEX.htm
- Click the View button to verify the link works.
- Click on the Save icon.

CME\7960 running SIP firmware - How do i setup incoming calls? - Can anyone help please?

Hi Guys,
I have a SIP trunk setup with a 2811 running CME version 7. I can make outbound calls ok but having issues getting the incoming calls working, i have 1 number on my SIP trunk and that is 01133501788 and i want that to ring my Cisco 7960 which is running SIP firmware not SCCP. I have included by config for anyone who can help me, i just want the incoming call to work.
Many Thanks.
Matthew.
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
   network 10.10.10.0 255.255.255.0
   default-router 10.10.10.1
   dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
   network 192.168.1.0 255.255.255.0
   default-router 192.168.1.1
   dns-server 188.92.232.50 188.92.232.100
   option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
bind control source-interface FastEthernet0/1.20
bind media source-interface FastEthernet0/1.20
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0008072514198272
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^9$.*$/ /\1/
voice translation-rule 2
rule 1 /^6...$/ /4143*002/
voice translation-profile DiscardDigit9
translate calling 2
translate called 1
voice translation-profile IncomingSIP
translate calling 1133501788
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing DiscardDigit9
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#

You my friend are a star! worked straight away, many thanks. Just one more thing, when i make an outgoing call, it always appears as "blocked" on my phone, my sip trunk is set to allow CME to alter outgoing CLI's how would i program the outgoing CLI to 01133501788 also?
The new working config is below with your suggestion, which works!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname Router
boot-start-marker
boot-end-marker
logging message-counter syslog
no aaa new-model
clock timezone GMT 0
dot11 syslog
ip source-route
ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.1.1
ip dhcp excluded-address 10.10.10.1
ip dhcp pool DATA_POOL
   network 10.10.10.0 255.255.255.0
   default-router 10.10.10.1
   dns-server 188.92.232.50 188.92.232.100
ip dhcp pool VOICE_POOL
   network 192.168.1.0 255.255.255.0
   default-router 192.168.1.1
   dns-server 188.92.232.50 188.92.232.100
   option 150 ip 192.168.1.1
ip name-server 188.92.232.50
ip name-server 188.92.232.100
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
registrar server
voice class codec 1
codec preference 2 g711ulaw
codec preference 3 g711alaw
voice register global
mode cme
source-address 192.168.1.1 port 5060
max-dn 144
max-pool 42
load 7960-7940 P0S3-8-12-00
authenticate register
tftp-path flash:
create profile sync 0015244443466064
voice register dn 1
number 6999
allow watch
name SIP
label SIP
voice register pool 1
id mac 000F.902B.40E0
type 7960
number 1 dn 1
dtmf-relay sip-notify
username cisco password cisco
codec g711ulaw
voice translation-rule 1
rule 1 /^6...$/ /4143*002/
voice translation-rule 3
rule 1 /^01133501788$/ /6999/
rule 2 /^1133501788$/ /6999/
voice translation-profile IncomingSIP
translate called 3
voice translation-profile Translatetrunk
translate calling 1
voice-card 0
no dspfarm
username matt privilege 15 secret 5 $1$DCD0$SjWqnKgDSGVzzIKRerXh11
archive
log config
hidekeys
interface FastEthernet0/0
ip address 194.12.0.222 255.255.255.252
ip nat outside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1
no ip address
ip nat inside
ip virtual-reassembly
duplex auto
speed auto
interface FastEthernet0/1.10
description DATA
encapsulation dot1Q 10
ip address 10.10.10.1 255.255.255.0
ip nat inside
ip virtual-reassembly
interface FastEthernet0/1.20
description VOICE
encapsulation dot1Q 20
ip address 192.168.1.1 255.255.255.0
ip nat inside
ip virtual-reassembly
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 194.12.0.221
ip http server
ip http authentication local
no ip http secure-server
ip nat inside source list 1 interface FastEthernet0/0 overload
access-list 1 permit 192.168.1.0 0.0.0.255
access-list 1 permit 10.10.10.0 0.0.0.255
tftp-server flash:P003-8-12-00.bin
tftp-server flash:P003-8-12-00.sbn
tftp-server flash:P0S3-8-12-00.loads
tftp-server flash:P0S3-8-12-00.sb2
tftp-server flash:P003-8-12-00
tftp-server flash:P003-8-12-00.loads
tftp-server flash:P003-8-12-00.sb2
tftp-server flash:SIP000F902B40E0.cnf.xml
control-plane
mgcp behavior g729-variants static-pt
dial-peer cor custom
dial-peer voice 1 voip
description IncomingSIP
translation-profile incoming IncomingSIP
voice-class codec 1
session protocol sipv2
session target sip-server
incoming called-number .T
dtmf-relay sip-notify rtp-nte
no vad
dial-peer voice 2 voip
description Outgoing Geographic
translation-profile outgoing Translatetrunk
destination-pattern 0[7]........
voice-class codec 1
session protocol sipv2
session target dns:sip.cloudcalling.co.uk
dtmf-relay rtp-nte
no vad
sip-ua
credentials username 4143*002 password 7 password realm sip.cloudcalling.co.uk
authentication username 4143*002 password 7 password
nat symmetric role passive
nat symmetric check-media-src
calling-info sip-to-pstn number set 4143*002
no remote-party-id
retry invite 3
retry register 3
timers connect 100
registrar dns:sip.cloudcalling.co.uk expires 60
sip-server dns:sip.cloudcalling.co.uk
host-registrar
gatekeeper
shutdown
telephony-service
load 7960-7940 P0S3-8-12-00
max-ephones 24
max-dn 30
ip source-address 192.168.1.1 port 2000
max-conferences 8 gain -6
web admin system name Admin secret 5 $1$Fktw$t9GQkdDdHmoYdwptO8.or.
transfer-system full-consult
create cnf-files version-stamp 7960 Dec 17 2013 14:35:13
line con 0
line aux 0
line vty 0 4
login
scheduler allocate 20000 1000
ntp server 85.119.80.232
end
Router#

Cisco SIP Phone 9971 won't register on CME 8.6 or 8.5 Please HELP

Please help me , I have problem with registering Cisco SIP phone 9971 with CME 8.6 on ISR 2901.
I configured CME for SIP clients, then I add configuration for 9971 phone and create profiles. Phone downloaded SEP...xml file from CME,after that phone look for g4-tones.xml and gd-sip.jar files, I added them to CME after that phone downloaded them and reboot. Now phone is stuck in some kind of loop and does not register on CME.
On phone log I can see repeting next few messeges.
12:01:58a No DNS Server IP
12:01:59a Updating Trust list
12:01:59a No Trust List instaled
12:01:59a SEP04C5AB03B0D.cnf.xml (TFTP) // at this time phone download SEP...xml file from CME
12:02:00a VPN Error: VPN is not Configured
on CME if issue DEBUG TFTP EVENTS i receive next few lines
*Aug 18 18:20:19.891: TFTP: Looking for CTLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:19.987: TFTP: Looking for ITLSEP04C5A4B03B0D.tlv
*Aug 18 18:20:20.083: TFTP: Looking for ITLFile.tlv
*Aug 18 18:20:20.347: TFTP: Looking for SEP04C5A4B03B0D.cnf.xml
*Aug 18 18:20:20.351: TFTP: Opened flash:/SEP04C5A4B03B0D.cnf.xml, fd 14, size 4585 for process 141
*Aug 18 18:20:20.363: TFTP: Finished flash:/SEP04C5A4B03B0D.cnf.xml, time 00:00:00 for process 141
here you can see verison info of CME
Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M, RELEASE SOFTWARE (fc1)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2011 by Cisco Systems, Inc.
Compiled Thu 24-Mar-11 15:31 by prod_rel_team
ROM: System Bootstrap, Version 15.0(1r)M9, RELEASE SOFTWARE (fc1)
ELTOSAN_ROUTER uptime is 1 hour, 50 minutes
System returned to ROM by reload at 16:29:20 UTC Thu Aug 18 2011
System image file is "flash:/c2900-universalk9-mz.SPA.151-4.M.bin"
Last reload type: Normal Reload
Last reload reason: Reload Command
Cisco CISCO2901/K9 (revision 1.0) with 471040K/53248K bytes of memory.
Processor board ID FGL1508252Y
3 Gigabit Ethernet interfaces
2 terminal lines
1 Virtual Private Network (VPN) Module
4 Voice FXO interfaces
4 Voice FXS interfaces
1 Internal Services Module (ISM) with Services Ready Engine (SRE)
   Survivable Remote Site Voicemail (SRSV) on Cisco Unity Express (CUE) 8.5.1 in slot/sub-slot 0/0
DRAM configuration is 64 bits wide with parity enabled.
255K bytes of non-volatile configuration memory.
254464K bytes of ATA System CompactFlash 0 (Read/Write)
License Info:
License UDI:
Device#   PID                   SN
*0        CISCO2901/K9          xxxxxxxxxxxxx
Technology Package License Information for Module:'c2900'
Technology    Technology-package          Technology-package
              Current       Type          Next reboot
ipbase        ipbasek9      Permanent     ipbasek9
security      securityk9    Permanent     securityk9
uc            uck9          Permanent     uck9
data          None          None          None
Configuration register is 0x2102
this is RUNNING CONFIGURATION
! Last configuration change at 16:10:12 UTC Thu Aug 18 2011
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname ELTOSAN_ROUTER
boot-start-marker
boot system flash:/c2900-universalk9-mz.SPA.151-4.M.bin
boot-end-marker
no aaa new-model
no ipv6 cef
ip source-route
no ip routing
no ip cef
no ip dhcp use vrf connected
ip dhcp excluded-address 192.168.5.1 192.168.5.10
ip dhcp excluded-address 192.168.5.200 192.168.5.255
ip dhcp pool phone
   network 192.168.5.0 255.255.255.0
   default-router 192.168.5.251
   option 150 ip 192.168.5.251
ip dhcp pool data
   relay source 192.168.2.0 255.255.255.0
   relay destination 192.168.2.201
multilink bundle-name authenticated
crypto pki token default removal timeout 0
voice-card 0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
fax protocol pass-through g711alaw
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 192.168.5.251 port 5060
max-dn 6
max-pool 6
load 9971 sip9971.9-1-1SR1.loads
authenticate register
tftp-path flash:
create profile sync 0005135312289902
voice register dn 1
number 207
allow watch
name GossaVM
label 207
voice register dn 3
number 101
name Dejan
label 101
mwi
voice register pool 1
id mac 000C.29C5.0011
number 1 dn 1
dtmf-relay sip-notify
username testvm password testera
codec g711alaw
voice register pool 3
id mac 04C5.A4B0.3B0D
type 9971
number 3 dn 3
presence call-list
dtmf-relay rtp-nte
username dejan password 1234
codec g711alaw
no vad
license udi pid CISCO2901/K9 sn xxxxxxxxxxxx
hw-module ism 0
hw-module pvdm 0/0
redundancy
interface GigabitEthernet0/0
description INTERFACE INTERNAL
no ip address
no ip route-cache
duplex auto
speed auto
no mop enabled
interface GigabitEthernet0/0.2
description LAN DATA
encapsulation dot1Q 2
ip address 192.168.2.251 255.255.255.0
no ip route-cache
interface GigabitEthernet0/0.5
description LAN VOICE
encapsulation dot1Q 5
ip address 192.168.5.251 255.255.255.0
no ip route-cache
interface ISM0/0
no ip address
no ip route-cache
shutdown
!Application: SRSV-CUE Running on ISM
interface GigabitEthernet0/1
no ip address
no ip route-cache
shutdown
duplex auto
speed auto
interface ISM0/1
description Internal switch interface connected to Internal Service Module
shutdown
interface Vlan1
no ip address
no ip route-cache
shutdown
ip forward-protocol nd
no ip http server
no ip http secure-server
snmp-server community public RO
tftp-server flash:dkern9971.100609R2-9-1-1SR1.sebn alias dkern9971.100609R2-9-1-1SR1.sebn
tftp-server flash:kern9971.9-1-1SR1.sebn alias kern9971.9-1-1SR1.sebn
tftp-server flash:rootfs9971.9-1-1SR1.sebn alias rootfs9971.9-1-1SR1.sebn
tftp-server flash:sboot9971.031610R1-9-1-1SR1.sebn alias sboot9971.031610R1-9-1-1SR1.sebn
tftp-server flash:skern9971.022809R2-9-1-1SR1.sebn alias skern9971.022809R2-9-1-1SR1.sebn
tftp-server flash:sip9971.9-1-1SR1.loads alias sip9971.9-1-1SR1.loads
tftp-server flash:United_States/g4-tones.xml
tftp-server flash:English_United_States/gd-sip.jar
control-plane
voice-port 0/0/0
voice-port 0/0/1
voice-port 0/0/2
voice-port 0/0/3
voice-port 0/1/0
voice-port 0/1/1
voice-port 0/1/2
voice-port 0/1/3
mgcp profile default
gatekeeper
shutdown
line con 0
line aux 0
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line vty 0 4
password jebiga
login
transport input all
end
I did not have any kind of problem with X-LITE to register to CME. also try with few SCCP phones 7940 and I did not any kind of problem .
this is content of SEP....xml file for 9971
<device>
<deviceProtocol>SIP</deviceProtocol>
<devicePool>
<dateTimeSetting>
<dateTemplate>M/D/YA</dateTemplate>
<timeZone>Pacific Standard/Daylight Time</timeZone>
<ntps>
<ntp priority="0">
<name>0.0.0.0</name>
<ntpMode>unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<sipPort>5060</sipPort>
</ports>
<processNodeName>192.168.5.251</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<sipProfile>
<sipProxies>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<localCfwdEnable>true</localCfwdEnable>
<callForwardURI>service-uri-cfwdall</callForwardURI>
<callPickupURI>service-uri-pickup</callPickupURI>
<callPickupGroupURI>service-uri-gpickup</callPickupGroupURI>
<callHoldRingback>2</callHoldRingback>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>2</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<remotePartyID>true</remotePartyID>
</sipStack>
<sipLines>
<line button="1" lineIndex="1">
<featureID>9</featureID>
<featureLabel></featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name></name>
<displayName></displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
<line button="2" lineIndex="2">
<featureID>9</featureID>
<featureLabel>101</featureLabel>
<proxy>USECALLMANAGER</proxy>
<port>5060</port>
<name>101</name>
<displayName>Dejan Rakic</displayName>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>1</callWaiting>
<authName>dejan</authName>
<authPassword>1234</authPassword>
<sharedLine>false</sharedLine>
<messagesNumber></messagesNumber>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>true</callerNumber>
<redirectedNumber>true</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<enableVad>true</enableVad>
<preferredCodec>g711alaw</preferredCodec>
<dialTemplate></dialTemplate>
<kpml>1</kpml>
<phoneLabel></phoneLabel>
<stutterMsgWaiting>2</stutterMsgWaiting>
<disableLocalSpeedDialConfig>true</disableLocalSpeedDialConfig>
<dscpForAudio>184</dscpForAudio>
<dscpVideo>136</dscpVideo>
</sipProfile>
<commonProfile>
<phonePassword>1234</phonePassword>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<featurePolicyFile>featurePolicyDefault.xml</featurePolicyFile>
<loadInformation>sip9971.9-1-1SR1.loads</loadInformation>
<vendorConfig>
</vendorConfig>
<commonConfig>
<videoCapability>0</videoCapability>
<ciscoCamera>0</ciscoCamera>
</commonConfig>
<sshUserId>dejan</sshUserId>
<sshPassword>1234</sshPassword>
<userId></userId>
<phoneServices>
<provisioning>2</provisioning>
<phoneService type="1" category="0">
<name>Missed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/MissedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Received Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/ReceivedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="1" category="0">
<name>Placed Calls</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/PlacedCalls</url>
<vendor></vendor>
<version></version>
</phoneService>
<phoneService type="2" category="0">
<name>Voicemail</name>
<phoneLabel></phoneLabel>
<url>Application:Cisco/Voicemail</url>
<vendor></vendor>
<version></version>
</phoneService>
</phoneServices>
<versionStamp>0131511014412102</versionStamp>
<userLocale>
<name>English_United_States</name>
<langCode>en</langCode>
</userLocale>
<networkLocale>United_States</networkLocale>
<networkLocaleInfo>
<name>United_States</name>
</networkLocaleInfo>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL>http://192.168.5.251:80/CMEserverForPhone/serviceurl</servicesURL>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
</device>

Hello,
I'm facing exactly the same problem, that is:
a Cisco SIP Phone 9971 won't register on CME 8.6 running on a 2811
I have read all the postings to this Forum, but I have not been able to solve it.
In my case the commands voice register dn and voice register pool are OK.
So frankly, I have no idea what I could be missing.
I'm pasting the Router's config.
I hope somebody is able to point me in the right direction.
Here is the config. Thank you!
C2811#sh run
Building configuration...
version 15.1
service timestamps debug datetime msec
service timestamps log datetime msec
no service password-encryption
hostname C2811
no aaa new-model
dot11 syslog
ip source-route
ip cef
ip dhcp excluded-address 172.25.140.1 172.25.140.10
ip dhcp excluded-address 172.35.140.1 172.35.140.10
ip dhcp pool Data
network 172.25.140.0 255.255.255.0
default-router 172.25.140.1
option 150 ip 172.25.140.1
dns-server 172.25.140.1
ip dhcp pool Voice
network 172.35.140.0 255.255.255.0
default-router 172.35.140.1
option 150 ip 172.35.140.1
dns-server 172.35.140.1
no ip domain lookup
no ipv6 cef
multilink bundle-name authenticated
voice service voip
allow-connections sip to sip
sip
registrar server expires max 3600 min 120
voice register global
mode cme
source-address 172.25.140.1 port 5060
max-dn 40
max-pool 42
load 9971 sip9971.9-4-1-9.loads
authenticate register
authenticate realm cisco
tftp-path flash:
create profile sync 0004820400584603
voice register dn 1
number 1010
allow watch
name Phone10
label Phone10
mwi
voice register pool 1
id mac 189C.5DB6.BD09
type 9971
number 1 dn 1
presence call-list
dtmf-relay rtp-nte
username adm password adm
call-forward b2bua busy 68600
codec g711ulaw
no vad
camera
video
voice-card 0
crypto pki token default removal timeout 0
crypto pki trustpoint TP-self-signed-1879153754
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-1879153754
revocation-check none
rsakeypair TP-self-signed-1879153754
crypto pki certificate chain TP-self-signed-1879153754
certificate self-signed 01
(details ommited)
license udi pid CISCO2811 sn FTX1146A44H
username admin privilege 15 password 0 admin
redundancy
interface FastEthernet0/0
no ip address
duplex auto
speed auto
interface FastEthernet0/0.25
description Data VLAN
encapsulation dot1Q 25
ip address 172.25.140.1 255.255.255.0
interface FastEthernet0/0.35
description Voice VLAN
encapsulation dot1Q 35
ip address 172.35.140.1 255.255.255.0
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 600 life 86400 requests 10000
tftp-server flash:P00308010200.bin
tftp-server flash:P00308010200.sbn
tftp-server flash:P00308010200.sb2
tftp-server flash:P00308010200.loads
tftp-server flash:SCCP42.9-3-1SR3-1S.loads
tftp-server flash:apps42.9-3-1ES19.sbn
tftp-server flash:cnu42.9-3-1ES19.sbn
tftp-server flash:cvm42sccp.9-3-1ES19.sbn
tftp-server flash:dsp42.9-3-1ES19.sbn
tftp-server flash:jar42sccp.9-3-1ES19.sbn
tftp-server flash:term42.default.loads
tftp-server flash:term62.default.loads
tftp-server flash:SCCP45.9-3-1SR3-1S.loads
tftp-server flash:apps45.9-3-1ES19.sbn
tftp-server flash:cnu45.9-3-1ES19.sbn
tftp-server flash:cvm45sccp.9-3-1ES19.sbn
tftp-server flash:dsp45.9-3-1ES19.sbn
tftp-server flash:jar45sccp.9-3-1ES19.sbn
tftp-server flash:term45.default.loads
tftp-server flash:term65.default.loads
tftp-server flash:/Ringtones/Ringlist.xml alias Ringlist.xml
tftp-server flash:/Ringtones/DistinctiveRingList.xml alias DistinctiveRingList.x
ml
tftp-server flash:sip9971.9-4-1-9.loads
tftp-server flash:kern9971.9-4-1-9.sebn
tftp-server flash:rootfs9971.9-4-1-9.sebn
tftp-server flash:dkern9971.100609R2-9-4-1-9.sebn
tftp-server flash:sboot9971.031610R1-9-4-1-9.sebn
tftp-server flash:skern9971.022809R2-9-4-1-9.sebn
tftp-server flash:/g4-tones.xml alias United_States/g4-tones.xml
tftp-server flash:/gd-sip.jar alias English_United_States/gd-sip.jar
control-plane
mgcp profile default
telephony-service
max-ephones 24
max-dn 48
ip source-address 172.25.140.1 port 2000
cnf-file location flash:
load 7960-7940 P00308010200
load 7942 SCCP42.9-3-1SR3-1S.loads
load 7945 SCCP45.9-3-1SR3-1S.loads
load 7962 SCCP42.9-3-1SR3-1S.loads
load 7965 SCCP45.9-3-1SR3-1S.loads
max-conferences 8 gain -6
dn-webedit
transfer-system full-consult
create cnf-files version-stamp 7960 Feb 11 2014 07:18:32
ephone-dn 1
number 1001
description Phone 1
name Phone 1
hold-alert 30 originator
ephone-dn 2
number 1002
description Phone 2
name Phone 2
hold-alert 30 originator
ephone-dn 3
number 1003
description Phone 3
name Phone 3
hold-alert 30 originator
ephone 1
device-security-mode none
mac-address 001C.58FB.6E0F
button 1:1
ephone 2
device-security-mode none
mac-address 0014.A981.7F8A
button 1:2
ephone 3
device-security-mode none
mac-address 0006.5356.A4B8
button 1:3
alias exec con conf t
alias exec sib show ip int brief
alias exec srb show run | b
alias exec sri show run int
line con 0
exec-timeout 0 0
logging synchronous
line aux 0
line vty 0 4
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
line vty 5 15
privilege level 15
login local
transport input telnet ssh
transport output telnet ssh
scheduler allocate 20000 1000
ntp master 1
end
C2811#

Help needed on Outbound Goods Receipt (IDoc)...

Hi Experts,
Can we use the message type WMMBXY and IDoc type WMMBID02 for Outbound Goods Receipts scenario also (Sending Goods Receipts information to target system).
Because WMMBID02 seems to be for Inbound going by the documentation in WE60. Can we use this for outbound as well? All the field mapping is perfect with respect to this IDoc and all the required fields exist in this IDoc.
I am struggling to find a suitable outbound process code and processing function module to trigger Idoc from MIGO Transaction once the goods receipt is created / generated.
Please advice me on how to achieve the above functionality.
Thanks in Advance.

Hi,
check with this link it might be helpful to u ...
<u>Purchase Order
Re: Problem to send idoc for a Good Receipt created
https://forums.sdn.sap.com/click.jspa?searchID=-1&messageID=1279261</u>
Hope this will surely help u ...
regards,
sana.
reward for useful answers...

CME 7.1 with SCCP 7940G phones and SIP connection to a VOIP provider - inbound outbound fails

Here's a quick and dirty diagram of a CME 7.1 configuration. The phone can all call each other but something is not quite right with the SIP provider. The registrar and SIP registration pieces are working but most of the configuration examples that I've seen make me think that the CME router was being used as the edge device to the internet. From my drawing, you can see that is not the case here. My edge device is a Cisco ASA5505 with 9.2.x software running. I might be missing something in the SIP gateway knowledge department. Without diving into the configuration, I'm wondering if SIP messages are failing for calls because of NAT'ing? Trying to do searches has been tricky because I keep running into information that is more about setting up CME for SIP phones or just getting SIP to work between CME and a SIP provider. I have that part working. I'm just a bit unsure about how an SCCP 7940G gets an outbound call or even gets one to come in.
When I dial from my cell phone to the pilot number, there are no rings, it just goes to the VOIP provider's voice mail. When I try to dial out, I get a fast busy.
So, is NAT a consideration? Will the SIP gateway set up a call (forward) via the pre-established SIP connection? Yeah, I do sound like a newb.
If anyone has good information about, let's say, an inbound call and how that traffic flow works.
Thanks!

Have you configured your ASA to either NAT the IP address of the CME router or to do port forwarding for port 5060?

Urgent help needed-Updating an outbound delivery.

Hi,
I have created an outbound delivery for a stock transport order in using VL10D.
The otbound delivery then goes to a different system wherein it is updated with picking,packing and shipping info.Then it comes back to SAP in the form of idoc(DELVRY05).Now what I need is to write a code in an exit(I know where to write the code) from where the data in the idoc will update the outbouned delivery with picked and delivered quantity.
I tried using FM's SD_DELIVERY_UPDATE_PICKING and WS_DELIVERY_UPDATE_2 but it is not working.Also where will I get to see the update if it happens.
Please help me.Its required urgently.
Thanks,
Sandeep.

Sandeep,
I m trying to understand this requirement. Your IDOC comes in with updated information for a delivery thats already available in the system.
In this case, the IDOC would lock the delivery object for its ude, so writing FM's directly might not help.
Please let me know your process code or your inbound FM. If it is IDOC_INPUT_DELVRY, then it already uses a WS_DELIVERY_UPDATE_2 , so there is little reason why you should be calling the FM again.
Please note that when this comes in it creates an inbound delivery rather an outbound. So, is your third party system sending you back an outbound IDOC?
Do you want to update the delivery that u sent out ( SD) to the third party system from that information.
In that case, please use BAPI_OUTB_DELIVERY_CONFIRM_DEC. This is for delivery conformation from a third party system. The documentation is very clear and there are toms of examples on SDN.
Please reward if useful.
Thanks
Ganesh.S

Urgent-Help needed in FM to update picking quantity in outbound delivery.

Hi,
I have to automatically update the delivery and picking quantity for an outbound delivery without doing any post goods issue.Could you please help me with any FM which does this with proper explanation.
Any help will be greatly appreciated.
Thanks,
Sandeep.

Check with FM : SD_DELIVERY_UPDATE_PICKING
The above FM will update Picking in delivery
Check the structure VBPOK ,within structure VBPOKKOMMI(Include structure)
Field is PIKMG.
Thanks
Seshu

Really Need Some Help with CME 8.6 using IOS as Firewall and Anyconnect VPN on Phones

Hello,
I have a 2911 Router with IOS Security and Voice enabled and we are using CME 8.6. I am using a built-in Anyconnect VPN on 3 phones that are for remote users and thus I needed to enable security zones on the router which works because the remote phones will boot up, get their phone configs and I am able to call those remote phones from an outside line.
The issue I am having is that when I try to dial a remote phone connected via the VPN through port g0/0 from and internal office phone, i.e., NOT involving the PSTN then there is no audio. It's as if no audio is going back and forth. When I take off the security zones from the virtual-template interface and the g0/0 interface then the audio works great and I can reach the phone from internal as I am supposed to.
Could someone take a peek at my security config and see why audio would not be traveling through the VPN when I have my security zones turned on?
clock timezone PST -8 0
clock summer-time PST recurring
network-clock-participate wic 0
network-clock-select 1 T1 0/0/0
no ipv6 cef
ip source-route
ip cef
ip dhcp excluded-address 192.168.8.1 192.168.8.19
ip dhcp pool owhvoip
network 192.168.8.0 255.255.248.0
default-router 192.168.8.1
option 150 ip 192.168.8.1
lease 30
multilink bundle-name authenticated
isdn switch-type primary-ni
crypto pki server cme_root
database level complete
grant auto
lifetime certificate 7305
lifetime ca-certificate 7305
crypto pki token default removal timeout 0
crypto pki trustpoint cme_root
enrollment url http://192.168.8.1:80
revocation-check none
rsakeypair cme_root
crypto pki trustpoint cme_cert
enrollment url http://192.168.8.1:80
revocation-check none
crypto pki trustpoint TP-self-signed-2736782807
enrollment selfsigned
subject-name cn=IOS-Self-Signed-Certificate-2736782807
revocation-check none
rsakeypair TP-self-signed-2736782807
voice-card 0
dspfarm
dsp services dspfarm
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
vpn-group 1
vpn-gateway 1 https://66.111.111.111/SSLVPNphone
vpn-trustpoint 1 trustpoint cme_cert leaf
vpn-profile 1
host-id-check disable
voice class codec 1
codec preference 1 g711ulaw
voice class custom-cptone jointone
dualtone conference
frequency 600 900
cadence 300 150 300 100 300 50
voice class custom-cptone leavetone
dualtone conference
frequency 400 800
cadence 400 50 200 50 200 50
voice translation-rule 1
rule 1 /9400/ /502/
rule 2 /9405/ /215/
rule 3 /9410/ /500/
voice translation-rule 2
rule 1 /.*/ /541999999/
voice translation-rule 100
rule 1 /^9/ // type any unknown plan any isdn
voice translation-profile Inbound_Calls_To_CUE
translate called 1
voice translation-profile InternationalType
translate called 100
voice translation-profile Local-CLID
translate calling 2
license udi pid CISCO2911/K9 sn FTX1641AHX3
hw-module pvdm 0/0
hw-module pvdm 0/1
hw-module sm 1
username routeradmin password 7 091649040910450B41
username cmeadmin privilege 15 password 7 03104803040E375F5E4D5D51
redundancy
controller T1 0/0/0
cablelength long 0db
pri-group timeslots 1-12,24
class-map type inspect match-any sslvpn
match protocol tcp
match protocol udp
match protocol icmp
class-map type inspect match-all router-access
match access-group name router-access
policy-map type inspect firewall-policy
class type inspect sslvpn
inspect
class class-default
drop
policy-map type inspect outside-to-router-policy
class type inspect router-access
inspect
class class-default
drop
zone security trusted
zone security internet
zone-pair security trusted-to-internet source trusted destination internet
service-policy type inspect firewall-policy
zone-pair security untrusted-to-trusted source internet destination trusted
service-policy type inspect outside-to-router-policy
interface Loopback0
ip address 192.168.17.1 255.255.248.0
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
description Internet
ip address dhcp
no ip redirects
no ip proxy-arp
zone-member security internet
duplex auto
speed auto
interface GigabitEthernet0/1
ip address 192.168.8.1 255.255.248.0
duplex auto
speed auto
interface GigabitEthernet0/2
no ip address
shutdown
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 Integrated-Service-Engine1/0
ip unnumbered Loopback0
service-module ip address 192.168.17.2 255.255.248.0
!Application: CUE Running on NME
service-module ip default-gateway 192.168.17.1
no keepalive
interface Virtual-Template1
ip unnumbered GigabitEthernet0/0
zone-member security trusted
ip local pool SSLVPNPhone_pool 192.168.9.1 192.168.9.5
ip forward-protocol nd
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:/cme-gui-8.6.0
ip route 192.168.17.2 255.255.255.255 Integrated-Service-Engine1/0
ip access-list extended router-access
permit tcp any host 66.111.111.111 eq 443
tftp-server flash:apps31.9-3-1ES26.sbn
control-plane
voice-port 0/0/0:23
voice-port 0/3/0
voice-port 0/3/1
mgcp profile default
sccp local GigabitEthernet0/1
sccp ccm 192.168.8.1 identifier 1 priority 1 version 7.0
sccp
sccp ccm group 1
bind interface GigabitEthernet0/1
associate ccm 1 priority 1
associate profile 1 register CME-CONF
dspfarm profile 1 conference
codec g729br8
codec g729r8
codec g729abr8
codec g729ar8
codec g711alaw
codec g711ulaw
maximum sessions 4
associate application SCCP
dial-peer voice 500 voip
destination-pattern 5..
session protocol sipv2
session target ipv4:192.168.17.2
dtmf-relay sip-notify
codec g711ulaw
no vad
dial-peer voice 10 pots
description Incoming Calls To AA
translation-profile incoming Inbound_Calls_To_CUE
incoming called-number .
port 0/0/0:23
dial-peer voice 20 pots
description local 10 digit dialing
translation-profile outgoing Local-CLID
destination-pattern 9[2-9].........
incoming called-number .
port 0/0/0:23
forward-digits 10
dial-peer voice 30 pots
description long distance dialing
translation-profile outgoing Local-CLID
destination-pattern 91..........
incoming called-number .
port 0/0/0:23
forward-digits 11
dial-peer voice 40 pots
description 911
destination-pattern 911
port 0/0/0:23
forward-digits all
dial-peer voice 45 pots
description 9911
destination-pattern 9911
port 0/0/0:23
forward-digits 3
dial-peer voice 50 pots
description international dialing
translation-profile outgoing InternationalType
destination-pattern 9T
incoming called-number .
port 0/0/0:23
dial-peer voice 650 pots
huntstop
destination-pattern 650
fax rate disable
port 0/3/0
gatekeeper
shutdown
telephony-service
protocol mode ipv4
sdspfarm units 5
sdspfarm tag 1 CME-CONF
conference hardware
moh-file-buffer 90
no auto-reg-ephone
authentication credential cmeadmin tshbavsp$$4
max-ephones 50
max-dn 200
ip source-address 192.168.8.1 port 2000
service dnis dir-lookup
timeouts transfer-recall 30
system message Oregon's Wild Harvest
url services http://192.168.17.2/voiceview/common/login.do
url authentication http://192.168.8.1/CCMCIP/authenticate.asp
cnf-file location flash:
cnf-file perphone
load 7931 SCCP31.9-3-1SR4-1S.loads
load 7936 cmterm_7936.3-3-21-0.bin
load 7942 SCCP42.9-3-1SR4-1S.loads
load 7962 SCCP42.9-4-2-1S.loads
time-zone 5
time-format 24
voicemail 500
max-conferences 8 gain -6
call-park system application
call-forward pattern .T
moh moh.wav
web admin system name cmeadmin secret 5 $1$60ro$u.0r/cno/OD2JmtvPq4w9.
dn-webedit
transfer-digit-collect orig-call
transfer-system full-consult
transfer-pattern .T
fac standard
create cnf-files version-stamp Jan 01 2002 00:00:00
ephone-template 1
softkeys connected Hold Park Confrn Trnsfer Endcall ConfList TrnsfVM
button-layout 7931 2
ephone-template 2
softkeys idle Dnd Gpickup Pickup Mobility
softkeys connected Hold Park Confrn Mobility Trnsfer TrnsfVM
button-layout 7931 2
ephone-dn 1 dual-line
number 200
label Lisa
name Lisa Ziomkowsky
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 2 dual-line
number 201
label Dylan
name Dylan Elmer
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 3 dual-line
number 202
label Kimberly
name Kimberly Krueger
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 4 dual-line
number 203
label Randy
name Randy Buresh
mobility
snr calling-number local
snr 915035042317 delay 5 timeout 15 cfwd-noan 500
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 5 dual-line
number 204
label Mark
name Mark McBride
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 6 dual-line
number 205
label Susan
name Susan Sundin
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 7 dual-line
number 206
label Rebecca
name Rebecca Vaught
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 8 dual-line
number 207
label Ronnda
name Ronnda Daniels
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 9 dual-line
number 208
label Matthew
name Matthew Creswell
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 10 dual-line
number 209
label Nate
name Nate Couture
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 11 dual-line
number 210
label Sarah
name Sarah Smith
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 12 dual-line
number 211
label Janis
name Janis McFerren
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 13 dual-line
number 212
label Val
name Val McBride
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 14 dual-line
number 213
label Shorty
name Arlene Haugen
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 15 dual-line
number 214
label Ruta
name Ruta Wells
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 16 dual-line
number 215
label 5415489405
name OWH Sales
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 17 dual-line
number 216
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 18 dual-line
number 217
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 19 dual-line
number 218
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 20 dual-line
number 219
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 21 dual-line
number 220
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 22 dual-line
number 221
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 23 dual-line
number 222
label Pam
name Pam Buresh
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 24 dual-line
number 223
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 25 dual-line
number 224
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 26 dual-line
number 225
label Elaine
name Elaine Mahan
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 27 octo-line
number 250
label Shipping
name Shipping
ephone-dn 28 dual-line
number 251
label Eli
name Eli Nourse
call-forward busy 500
call-forward noan 500 timeout 10
ephone-dn 29 dual-line
number 252
ephone-dn 30 dual-line
number 253
ephone-dn 31 octo-line
number 100
label Customer Service
name Customer Service
call-forward busy 500
call-forward noan 500 timeout 12
ephone-dn 32 octo-line
number 101
label Sales
name Sales
call-forward busy 214
call-forward noan 214 timeout 12
ephone-dn 33 dual-line
number 260
label Conference Room
name Conference Room
call-forward busy 100
call-forward noan 100 timeout 12
ephone-dn 100
number 300
park-slot timeout 20 limit 2 recall
description Park Slot For All Company
ephone-dn 101
number 301
park-slot timeout 20 limit 2 recall
description Park Slot for All Company
ephone-dn 102
number 302
park-slot timeout 20 limit 2 recall
description Park Slot for All Company
ephone-dn 103
number 700
name All Company Paging
paging ip 239.1.1.10 port 2000
ephone-dn 104
number 8000...
mwi on
ephone-dn 105
number 8001...
mwi off
ephone-dn 106 octo-line
number A00
description ad-hoc conferencing
conference ad-hoc
ephone-dn 107 octo-line
number A01
description ad-hoc conferencing
conference ad-hoc
ephone-dn 108 octo-line
number A02
description ad-hoc conferencing
conference ad-hoc
ephone 1
device-security-mode none
mac-address 001F.CA34.88AE
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:2 2:31
ephone 2
device-security-mode none
mac-address 001F.CA34.8A03
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:12
ephone 3
device-security-mode none
mac-address 001F.CA34.898B
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
ephone 4
device-security-mode none
mac-address 001F.CA34.893F
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
ephone 5
device-security-mode none
mac-address 001F.CA34.8A71
ephone-template 1
max-calls-per-button 2
username "susan"
paging-dn 103
type 7931
button 1:6
ephone 6
device-security-mode none
mac-address 001F.CA34.8871
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:7 2:31 3:32
ephone 7
device-security-mode none
mac-address 001F.CA34.8998
ephone-template 1
max-calls-per-button 2
username "matthew"
paging-dn 103
type 7931
button 1:9
ephone 8
device-security-mode none
mac-address 001F.CA36.8787
ephone-template 1
max-calls-per-button 2
username "nate"
paging-dn 103
type 7931
button 1:10
ephone 9
device-security-mode none
mac-address 001F.CA34.8805
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:5
ephone 10
device-security-mode none
mac-address 001F.CA34.880C
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:14
ephone 11
device-security-mode none
mac-address 001F.CA34.8935
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:3
ephone 12
device-security-mode none
mac-address 001F.CA34.8995
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:8 2:31
ephone 13
device-security-mode none
mac-address 0021.5504.1796
ephone-template 2
max-calls-per-button 2
paging-dn 103
type 7931
button 1:4
ephone 14
device-security-mode none
mac-address 001F.CA34.88F7
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:23
ephone 15
device-security-mode none
mac-address 001F.CA34.8894
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:26
ephone 16
device-security-mode none
mac-address 001F.CA34.8869
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:28 2:27
ephone 17
device-security-mode none
mac-address 001F.CA34.885F
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:11
ephone 18
device-security-mode none
mac-address 001F.CA34.893C
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:27
ephone 19
device-security-mode none
mac-address 001F.CA34.8873
ephone-template 1
max-calls-per-button 2
paging-dn 103
type 7931
button 1:27
ephone 20
device-security-mode none
mac-address A456.3040.B7DD
paging-dn 103
type 7942
vpn-group 1
vpn-profile 1
button 1:13
ephone 21
device-security-mode none
mac-address A456.30BA.5474
paging-dn 103
type 7942
vpn-group 1
vpn-profile 1
button 1:15 2:16 3:32
ephone 22
device-security-mode none
mac-address A456.3040.B72E
paging-dn 103
type 7942
vpn-group 1
vpn-profile 1
button 1:1
ephone 23
device-security-mode none
mac-address 00E0.75F3.D1D9
paging-dn 103
type 7936
button 1:33
line con 0
line aux 0
line 2
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
stopbits 1
line 67
no activation-character
no exec
transport preferred none
transport input all
transport output pad telnet rlogin lapb-ta mop udptn v120 ssh
line vty 0 4
transport input all
scheduler allocate 20000 1000
ntp master
ntp update-calendar
ntp server 216.228.192.69
webvpn gateway sslvpn_gw
ip address 66.111.111.111 port 443
ssl encryption 3des-sha1 aes-sha1
ssl trustpoint cme_cert
inservice
webvpn context sslvpn_context
ssl encryption 3des-sha1 aes-sha1
ssl authenticate verify all
policy group SSLVPNphone
functions svc-enabled
hide-url-bar
svc address-pool "SSLVPNPhone_pool" netmask 255.255.248.0
svc default-domain "bendbroadband.com"
virtual-template 1
default-group-policy SSLVPNphone
gateway sslvpn_gw domain SSLVPNphone
authentication certificate
ca trustpoint cme_root
inservice
end

I think your ACL could be the culprit.
ip access-list extended router-access
permit tcp any host 66.111.111.111 eq 443
Would you be able to change the entry to permit ip any any (just for testing purpose) and then test to see if the calls function properly. If they work fine then we know that we need to open som ports there.
Please remember to select a correct answer and rate helpful posts

Urgent help- How to open Outbound queues

Hi,
I am new to SCM-APO, we have SCM 5.0 having CIF to R/3 4.7 Enterprise.
In SMQ1 i can see lot of outbound queues present with status NOSEND in R/3.
I want to push these queue manually how shoild i do it
Please guide me as i m new to this area.
Thanks
Amit

Hi Amit,
Go to Transaction /n/sapapo/ccr i.e.CIF Comparision/Reconcillation of Transaction Data. then Click on
Execute Comparision/Reconcillation Tab. Then Enter Parter System i.e ECC,then Proudct & Location.then Select Document to be checked like Production Order i.e Mfg Ordrs(with Reqs/Receipt , with Opernations),Planned Order,Purchase order etc. which you want to Transfer in ECC concern to the Product.Then Execute the Transaction.
You will find Transaction data & result Documnet.Open the Transaction Data i.e Mfg Orders or Planned Orders.Then double click on Requirements/Receipts or Operations.You will find the objects in right hand side.Select all the objects & push in to R/3 with Push button R/3.
Hope so it will help you to solve your problem.
Regards
Sujay

HELP!!! - EDI Outbound HTTP call failure

Our EDI outbound (HTTPS-OXTA) is failing since Monday in production. We narrowed down the area that might be an issue. This is what we see in the Apache log,
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(469): OPM: EW: Enters opm_ew_broadcast()
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(517): OPM: EW: Broadcasts msg: cmd=Broadcast&<serverName>&8001&1337662614&JServ&DiscoGroup&<server_url>&1&1&0&31490&17001;FormsGroup&<server_url>&1&1&0&31491&18001;OACoreGroup&<server_url>&1&1&0&31489&16001;XmlSvcsGrp&<server_url>&1&1&0&31492&19001
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(291): OPM:hc: Connecting to url: <server_url>:8101/oprocmgr-service
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(314): OPM:hc: Connection to host: <server_url>, port: 8101
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(438): OPM:hc: HTTP Request sent to server: POST /oprocmgr-service<server_url> HTTP/1.1^M
Host: <server_url>^M
Content-Type: application/x-www-form-urlencoded^M
Content-Length: 269^M
cmd=Broadcast&<serverName>&8001&1337662614&JServ&DiscoGroup&<server_url>&1&1&0&31490&17001;FormsGroup&<server_url>&1&1&0&31491&18001;OACoreGroup&<server_url>&1&1&0&31489&16001;XmlSvcsGrp&<server_url>&1&1&0&31492&19001
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[0] is HTTP/1.1 404 Not Found
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[1] is Date: Tue, 22 May 2012 05:31:56 GMT
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[2] is Transfer-Encoding: chunked
[Tue May 22 01:31:56 2012] [debug] opm_hc.c(808): OPM:hc: headers[3] is Content-Type: text/html; charset=iso-8859-1
[Tue May 22 01:31:56 2012] [debug] opm_ew.c(525): OPM: EW: Broadcasts to <server_url> and send result=404
I'm trying to understand the steps of the process. Does "HTTP/1.1 404 Not Found" response to the opm_hc.c(438) call? When I type "<server_url>:8101" in the browser, I get "The webpage cannot be displayed" error. Does this should work?
EDI outbound is routed to proxy and confirmed that call from OTA was never made to proxy. Switched protocol to SMTP and it worked. There is no issue other than HTTP initial call failure. Any help you can give me I'd appreciated.

George great support so far (+5)
Hi Robert
debug ccsip all is very intensive so you should do the following before enabling the debug
service sequence-numbers
service timestamps debug datetime localtime msec
logging buffered 10000000 debug
no logging console
no logging monitor
default logging rate-limit
default logging queue-limit
Then..
<Enable debugs, then test again.>
debug ccsip all
<Enable session capture to txt file in terminal program.> (such as Putty)
then do the ff:
terminal length 0
show logging
++++
What is even more strange is that the call appears to be disconnected from the far end. From the logs below the outbound call leg (45) is where the disconnect is coming from and the "cc_api_call_disconnected" shows this call leg talking to CCAPI..
001858: *Jan 20 13:18:19.102: //45/8B56ECEE8011/CCAPI/cc_api_call_disconnected:
Cause Value=16, Interface=0x3CE6D670, Call Id=45
001859: *Jan 20 13:18:19.102: //45/8B56ECEE8011/CCAPI/cc_api_call_disconnected:
Call Entry(Responsed=TRUE, Cause Value=16, Retry Count=0)
Can you also send a debug voip ccapi onout from the CUBE. we need to check if the call arrives there, though we don't see any INVITE request sent out.

Error in outbound delv wrt MTO .. pl help.

Dear All ,
Uswer had made a M.T.O , & now he hasd created an outbound delv , now he is displaying in VL03N, here an info displays in task bar as " Check the notes in error log --> as he click on edit on menu tab on the top, the error with green signal is for all five line items as " Iten 0010mat staging 29-07-2008 replaced by current date pl guide that is this error will create any problem in invoicing or not as this error is with green color instead of red color , pl help
regards
sap11

ok

CME 4FXO Outbound Help

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