1. Configure DSP as SCCP device and register to CME.
voice-card 0
no dspfarm
dsp service dspfarm
sccp local voice-interface
sccp ccm cme-address identifier 1 version 7.0+
sccp
sccp ccm group 1
associate ccm 1 priority 1
associate profile 1 register CONF
dspfarm profile 1
associate application sccp
maximum session 2
no shut
telephony-service
sdspfarm unit 2
sdspfarm tag 1 CONF
conference hardware
2. Configure ephone-dn for Ad-hoc and Meet-me conference
ephone-dn 200 octo-line
number 6666
conference ad-hoc
ephone-dn 201 octo-line
number 7777
conference meetme
3. Change ephone-template to include conf button for connected status.
Tuesday, April 27, 2010
Monday, April 26, 2010
CME Night Service
The instructions from CME system admin guide is not complete when configuring Night Service. Below is a brief guide on how to enable the feature on CME.
1. The first step is to define night service hours under telephony-service configuration mode.
telephony-service
night-service day Sun 18:00 23:59
night-service day ...
2. The second step is to utilize night service.
ephone-dn 1
call-forward night-service ....
1. The first step is to define night service hours under telephony-service configuration mode.
telephony-service
night-service day Sun 18:00 23:59
night-service day ...
2. The second step is to utilize night service.
ephone-dn 1
call-forward night-service ....
Wednesday, September 9, 2009
CUE Licensing
CUE has two types of licensing, the license for core application, and the license for IVR.
For core application license, you can choose between CME and CM. You can have either CME or CM, but not both.
Difference between H245 Signal and H245 Alphanumaric
H245 Signal and H245 Alphanumaric are two DTMF relay methods for h.323 dial peers. They are both out of band DTMF relay methods.
H245 Signal transmits two events, the start and the stop when holding the key and releasing the key. In another word, it records the duration. It is useful in calling card scenario.
H245 Alphanumaric only sends one event, and it has the default during period for 200ms.
The Cisco recommended DTMF relay method is H245 Alphanumaric.
Thursday, August 20, 2009
CUE Upgrade from Version 1 to Version 7.0.3
I just bought a Cisco Unity Express module from ebay (NM-CUE), and it arrived this morning. I never touched it before. I connected it to a Cisco 3845 ISR.
After loading the IOS, I issued 'show version', and there is a service engine found. Using 'show ip interface brief' displays the new service engine interface.
So I configured an IP address on the service engine interface, and then under service engine interface configuration mode, I configured another ip address within same subnet of the service engine interface, pointing the default gateway address as the service engine interface address, and then enabled the interface.
Then I issued 'service-module service-engine slot/port session' command to switch the console to NM-CUE. Using 'show software version', I found out the version of CUE is 1.0.7. The version required on CCIE Voice blueprint is version 7.
Further investigations show that there are several ways for the upgrade, but after some testing, I realize that to upgrade from version 1 to version 7, the only way is to upgrade through boot helper.
Before doing the upgrade, some preparation work needs to be done. First is to download the related installation pack, language files and licenses for version 7 (I downloaded version 7.0.3).
FTP server and TFTP server need to be setup. TFTP is used to upgrade the boot helper, and FTP server is used to upgrade the software, language files and licenses. I simply put all files under one directory and point both ftp and tftp path to this directory.
Reload the CUE and type '***' to get into boot helper when it is prompted to do so.
Input 'config', and configure the related information, like ip address, subnet mask, default gateway, tftp server address and boot helper file name. When config is finished, type 'boot helper', and the boot helper upgrade is started. When finished, choose '1' to install software, and type the related package name, ftp server address, username and password. When prompted, select the language file to install.
When it is done, you can have some basic initialization configurations.
Using 'software install clean url' command to install the license files.
:-D I now have version 7.0.3 installed on Unity Express.
After loading the IOS, I issued 'show version', and there is a service engine found. Using 'show ip interface brief' displays the new service engine interface.
So I configured an IP address on the service engine interface, and then under service engine interface configuration mode, I configured another ip address within same subnet of the service engine interface, pointing the default gateway address as the service engine interface address, and then enabled the interface.
Then I issued 'service-module service-engine slot/port session' command to switch the console to NM-CUE. Using 'show software version', I found out the version of CUE is 1.0.7. The version required on CCIE Voice blueprint is version 7.
Further investigations show that there are several ways for the upgrade, but after some testing, I realize that to upgrade from version 1 to version 7, the only way is to upgrade through boot helper.
Before doing the upgrade, some preparation work needs to be done. First is to download the related installation pack, language files and licenses for version 7 (I downloaded version 7.0.3).
FTP server and TFTP server need to be setup. TFTP is used to upgrade the boot helper, and FTP server is used to upgrade the software, language files and licenses. I simply put all files under one directory and point both ftp and tftp path to this directory.
Reload the CUE and type '***' to get into boot helper when it is prompted to do so.
Input 'config', and configure the related information, like ip address, subnet mask, default gateway, tftp server address and boot helper file name. When config is finished, type 'boot helper', and the boot helper upgrade is started. When finished, choose '1' to install software, and type the related package name, ftp server address, username and password. When prompted, select the language file to install.
When it is done, you can have some basic initialization configurations.
Using 'software install clean url' command to install the license files.
:-D I now have version 7.0.3 installed on Unity Express.
Saturday, July 18, 2009
SIP Significant Digits - SigDigits
Configuration of Unified CVP SIP Service:
The Unified CVP SIP Service does not currently have a configuration field for setting the significant digits that should be stripped. Instead, you must edit the sip.properties file. The sip.properties file is located in the C:\Cisco\CVP\conf directory by default. Add the following line to the end of the sip.properties file (to strip four digits from the DNIS number):
SIP.SigDigits = 4
Configuration of VoiceXML gateway:
Configure the VXML gateway to match the DNIS string, including the prepended digits:
dial-peer voice 3000 voip
incoming-called number 33335551000T
service bootstrap
...
Configure the Unified CVP bootstrap.tcl application with the sigdigits parameter, telling it how many digits to strip off of the incoming DNIS string:
application
service bootstrap flash:bootstrap.tcl
param sigdigits 4
...
Cisco Unified CM configuration (if used):
Configure Unified CM to strip the prepended digits, either by using the Significant Digits configuration on the SIP Trunk configuration page or by using translation patterns.
SIP Proxy configuration:
Define static routes on the SIP Proxy, with the prepended digit present, to be sent to the appropriate VoiceXML gateway. Because transfers to agents on a Unified CM cluster will also have the digits prepended, the static routes for agent phones must also contain the prepended digits.
The Unified CVP SIP Service does not currently have a configuration field for setting the significant digits that should be stripped. Instead, you must edit the sip.properties file. The sip.properties file is located in the C:\Cisco\CVP\conf directory by default. Add the following line to the end of the sip.properties file (to strip four digits from the DNIS number):
SIP.SigDigits = 4
Configuration of VoiceXML gateway:
Configure the VXML gateway to match the DNIS string, including the prepended digits:
dial-peer voice 3000 voip
incoming-called number 33335551000T
service bootstrap
...
Configure the Unified CVP bootstrap.tcl application with the sigdigits parameter, telling it how many digits to strip off of the incoming DNIS string:
application
service bootstrap flash:bootstrap.tcl
param sigdigits 4
...
Cisco Unified CM configuration (if used):
Configure Unified CM to strip the prepended digits, either by using the Significant Digits configuration on the SIP Trunk configuration page or by using translation patterns.
SIP Proxy configuration:
Define static routes on the SIP Proxy, with the prepended digit present, to be sent to the appropriate VoiceXML gateway. Because transfers to agents on a Unified CM cluster will also have the digits prepended, the static routes for agent phones must also contain the prepended digits.
Friday, July 17, 2009
IOS Gateway Configuration for SIP-Proxy Redundancy
With Cisco IOS gateways, dial-peers are used to match phone numbers, and the destination can be a SIP Proxy Server, DNS SRV, or IP address. The following example shows a Cisco IOS gateway configuration to send calls to a SIP Proxy Server using the SIP Proxy's IP address.
sip-ua
sip-server ipv4:10.4.1.100:5060
dial-peer voice 1000 voip
session target sip-server
...
The sip-server command on the dial-peer tells the Cisco IOS gateway to use the globally defined
sip-server that is configured under the sip-ua settings. In order to configure multiple SIP Proxies for redundancy, you can change the IP address to a DNS SRV record, as shown in the following example. The DNS SRV record allows a single DNS name to be mapped to multiple servers.
sip-ua
sip-server dns:cvp.cisco.com
dial-peer voice 1000 voip
session target sip-server
...
Alternatively, you can configure multiple dial-peers to point directly at multiple SIP Proxy servers, as shown in the following example. This configuration allows you to specify IP addresses instead of relyingon DNS.
dial-peer voice 1000 voip
session target ipv4:10.4.1.100
preference 1
...
dial-peer voice 1000 voip
session target ipv4:10.4.1.101
preference 1
...
In the preceding examples, the calls are sent to the SIP Proxy server for dial plan resolution and callrouting. If there are multiple Unified CVP Call Servers, the SIP Proxy server would be configured with multiple routes for load balancing and redundancy.
DNS SRV records allow an administrator to configure redundancy and load balancing with finer
granularity than with DNS round-robin redundancy and load balancing. A DNS SRV record allows you to define which hosts should be used for a particular service (the service in this case is SIP), and it allowsyou to define the load-balancing characteristics among those hosts. In the following example, the
redundancy provided by the three dial-peers configured above is replaced with a single dial-peer using a DNS SRV record. Note that a DNS server is required in order to do the DNS lookups.
ip name-server 10.4.33.200
dial-peer voice 1000 voip
session target dns:cvp.cisco.com
With Cisco IOS gateways, it is possible to define DNS SRV records statically, similar to static host records. This capability allows you to simplify dial-peer configuration while also providing DNS SRV load balancing and redundancy. The downside of this method is that, if the SRV record needs to be changed, it must be changed on each gateway instead of on a centralized DNS server. The following example shows the configuration of static SRV records for SIP services handled by cvp.cisco.com, and the SIP SRV records for cvp.cisco.com are configured to load-balance across three servers.
ip host cvp4cc2.cisco.com 10.4.33.132
ip host cvp4cc3.cisco.com 10.4.33.133
ip host cvp4cc1.cisco.com 10.4.33.131
(SRV records for SIP/TCP)
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc3.cisco.com
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc2.cisco.com
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc1.cisco.com
(SRV records for SIP/UDP)
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc3.cisco.com
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc2.cisco.com
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc1.cisco.com
Cisco highly recommends the use of UDP instead of TCP for SIP signaling. TCP stack timeout delays can cause significant delays to the caller during failures.
sip-ua
sip-server ipv4:10.4.1.100:5060
dial-peer voice 1000 voip
session target sip-server
...
The sip-server command on the dial-peer tells the Cisco IOS gateway to use the globally defined
sip-server that is configured under the sip-ua settings. In order to configure multiple SIP Proxies for redundancy, you can change the IP address to a DNS SRV record, as shown in the following example. The DNS SRV record allows a single DNS name to be mapped to multiple servers.
sip-ua
sip-server dns:cvp.cisco.com
dial-peer voice 1000 voip
session target sip-server
...
Alternatively, you can configure multiple dial-peers to point directly at multiple SIP Proxy servers, as shown in the following example. This configuration allows you to specify IP addresses instead of relyingon DNS.
dial-peer voice 1000 voip
session target ipv4:10.4.1.100
preference 1
...
dial-peer voice 1000 voip
session target ipv4:10.4.1.101
preference 1
...
In the preceding examples, the calls are sent to the SIP Proxy server for dial plan resolution and callrouting. If there are multiple Unified CVP Call Servers, the SIP Proxy server would be configured with multiple routes for load balancing and redundancy.
DNS SRV records allow an administrator to configure redundancy and load balancing with finer
granularity than with DNS round-robin redundancy and load balancing. A DNS SRV record allows you to define which hosts should be used for a particular service (the service in this case is SIP), and it allowsyou to define the load-balancing characteristics among those hosts. In the following example, the
redundancy provided by the three dial-peers configured above is replaced with a single dial-peer using a DNS SRV record. Note that a DNS server is required in order to do the DNS lookups.
ip name-server 10.4.33.200
dial-peer voice 1000 voip
session target dns:cvp.cisco.com
With Cisco IOS gateways, it is possible to define DNS SRV records statically, similar to static host records. This capability allows you to simplify dial-peer configuration while also providing DNS SRV load balancing and redundancy. The downside of this method is that, if the SRV record needs to be changed, it must be changed on each gateway instead of on a centralized DNS server. The following example shows the configuration of static SRV records for SIP services handled by cvp.cisco.com, and the SIP SRV records for cvp.cisco.com are configured to load-balance across three servers.
ip host cvp4cc2.cisco.com 10.4.33.132
ip host cvp4cc3.cisco.com 10.4.33.133
ip host cvp4cc1.cisco.com 10.4.33.131
(SRV records for SIP/TCP)
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc3.cisco.com
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc2.cisco.com
ip host _sip._tcp.cvp.cisco.com srv 1 50 5060 cvp4cc1.cisco.com
(SRV records for SIP/UDP)
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc3.cisco.com
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc2.cisco.com
ip host _sip._udp.cvp.cisco.com srv 1 50 5060 cvp4cc1.cisco.com
Cisco highly recommends the use of UDP instead of TCP for SIP signaling. TCP stack timeout delays can cause significant delays to the caller during failures.
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