Cisco UCM CAR (CDR) Web GUI Access Request (https:///car/)

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/11_5_1_SU1/Administration/cucm_b_administration-guide-1151su1/cucm_b_administration-guide-1151su1_chapter_010.pdf

 

To provide a user to CAR (CDR)  (https://<CAR server IP Address>/car/) web page, the following two access groups must be associated with the user. After giving user this access, please test login to other areas of UCM GUI, so the users do not gain unapproved access to UCM Admin pages.

 

  1. Standard CCM End Users
  2. Standard Admin Rep Tool Admin = (Standard CAR Admin Users, Standard CCM Super Users)

 

CAR web gui

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CIPT2: 300-075 Taming the beast and my study note

Wow, what an experience it was, trying to pass Cisco CIPT2 300-075 exam during the last 5 weeks. So much was on the line as if I did not pass this exam by 17th of June, 2016, all my CCNP R&S and CCNP Voice was expiring, so I would be facing 7 exams to re-certify as CCNP in both technologies. Unfortunately, I had to tackle it 3 times to pass this exam, and got lucky on 3rd attempt. My first attempt was a lame attempt as I failed by 7 questions, the second attempt was a little bit more decent, failing by a SINGLE question. The passing mark for CIPT2 300-075 exam is 860/1000, which makes one question valued somewhere between 13 pts to 17 pts depending on the weight of the question Cisco is throwing at you. But today, I passed the exam and what a Roller Coaster ride this exam was, I’ve passed the exam with the exact passing score! Finally, the lady luck is on my side. After two failures, trying to tame the beast, I’ve studied so many hours trying to understand the VCS components and finally got a full grasp of the concept and basic configuration. Over 3 days of long weekend, I cranked out 30 hours of study time for VCS C and E studies (no pain, no gain! I am thankful that I’ve failed the second time by one question. I was forced to try my best). Oh, what a feeling!

 

I want to share some of my study notes with you so, you don’t have to do it the hard way like me, but I urge you to spend some time reading Cisco documentations, watch videos from CiscoLive and read the official study books front-to-end before jumping into the full study drive mode. I hope my notes will help someone on their way to becoming a CCNP-Collaboration. My notes are based on Cisco documents but also comes from the live environment and my experience, so it might not be 100%, but if you disagree with me on those questions, then show me your proof that you are in the right and I am in the wrong with a live Cisco documentations referencing the page and line number. As always, if you cannot avoid it, try to face it with a dignity or try to enjoy it!!! I would choose the latter…. 🙂

 

 

1. Regional configuration of Cisco VoIP environment
Note: Cisco Best practice, (G.729/24K) to compress BW for regions. Hardware MTP only supports G.711 a-law and G.711 u-law. Also regions will need transcoders if multiple codecs are deployed, NOT hardware MTP.
2. While using Query wizard to configure the trace and log central feature to collect install logs.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/service/7_1_2/rtmt/RTMT/rttlc.html
“The time zone of the client machine provides the default setting for the Select Reference Server Time Zone field. All the standard time zones, along with a separate set of entries for all time zones that have Daylight Saving settings, display in the Select Time Zone drop-down list box.”
“Trace and Log Central downloads the file with a time range that is based on your Selected Reference Server Time Zone field. If you have servers in a cluster in a different time zone, TLC will adjust for the time change and get files for the same period of time. For example, if you specify files from 9:00 AM to 10:00 AM and you have a second server (server x) that is in a time zone that is one hour ahead, TLC will download files from 10:00 AM to 11:00 AM from server x.”
3. Standardization of caller addresses between H.323 and SIP endpoints.
http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_guide_chapter09186a00 800eadee.htmlhttp://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-2/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-2.pdf (Page 17)
“The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices. ”
“The following transform modifies the destination alias of all call attempts made to destination aliases which do not contain an ‘@’. The old destination alias has @example.com appended to it. This has the effect of standardizing all called destination aliases into a SIP URI format.”
From VCS and CUCM Deployment guide:
“Thus, a transform is needed to ensure that the dialed number is transformed into a consistent form, in this case to add the domain (vcs.domain) if required.”
4. CUCM Extension Mobility characteristics
http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_administration_guide_chapter09186a0080153e60.html#wp1092734

“Able to adopt a user profile even when no user is logged in”
“Almost same attributes as a physical device”
5. A globalized dial plan, 3 ways enabling ingress gateways to process calls.

Configure the called-party transformation settings for incoming calls on H.323 gateways.
Configure translation patterns in the partitions used by the gateway calling search space
Configure the gateway with prefix digits to add necessary country and region codes.

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab10/collab10/dialplan.html#16685
“Localized Call Ingress on Gateways
The called and calling numbers delivered into the Unified Communications system by external networks (for example, the PSTN) are typically localized. The form of the numbers may vary, depending on the service provider’s configuration of the trunk. As a gateway is connected to a PSTN trunk, the system administrator must work with the PSTN service provider to determine the applicable signaling rules to be used for this specific trunk. As calls are delivered into the system from the trunk, some of the information about the calling and called numbers will be provided explicitly and some of it will be implied. Using this information, the system must derive the calls’ globalized calling and called party numbers.
The globalization of the called party number can be implemented through one of the following methods:
In the gateway configuration, configure Call Routing Information > Inbound Calls, where the quantity of significant digits to be retained from the original called number and the prefix digits to be added to the resulting string are used to globalize the called number. The prefix digits should be used to add the applicable + sign and country, region, and city codes.
Place translation patterns in partitions referenced by the gateway’s calling search space. The translation patterns should be configured to match the called party number form used by the trunks connected to the gateway, and should translate it into the global form. The prefix digits should be used to add the applicable + sign and country, region, and city codes.
Use the incoming call’s called party transformation settings available on the gateway and on the gateway’s device pool. There you can define strip and prefix digit instructions or alternatively configure a called party transformation calling search space per numbering type.
The globalization of the calling party number should be implemented by using the Incoming Calling Party Settings configured either on the gateway directly or in the device pool controlling the gateway.”
6. 2 types of devices are affected when an engineer changes the DSCP for Video Calls service parameter
Read “Set DSCP Values”.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/jabber/10_6/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide/CJAB_BK_C56DE1AB_00_cisco-jabber-106-deployment-and-installation-guide_appendix_01111.html
7. Cisco VCS uses 3 Presence status of endpoints for monitoring
http://www.cisco.com/c/en/us/td/docs/telepresence/infrastructure/articles/vcs_monitors_presence_ status_endpoints_kb_186.html
in-all
call-ended
registration
8. 3 steps configure Cisco Unified Survivable Remote Site Telephony for SIP phones

1. configure an SRST reference
2. Configure the SIP registrar
3. Configure voice register pool

 

 

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip

sip

registrar server expires max 600 min 60

!

voice register pool 10
id network 172.26.10.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp sip-notify
codec g711ulaw
no vad
!
voice register pool 11
id network 172.26.11.0 mask 255.255.255.0
dtmf-relay rtp-nte cisco-rtp sip-notify
codec g711ulaw
no vad
!

sip-ua
registrar ipv4:172.26.10.240 expires 600

(172.26.10.240 is the SRST gateway IP address).

Don’t forget on UCM SRST reference configuration for gateway:

On UCM SRST reference configuration for gateway

SIP Network/IP Address 172.26.10.240

SIP Port 5060
9. Device Mobility – overlapping parameters for roaming
Location
Network Locale
MRGL

Reference: https://supportforums.cisco.com/document/77096/device-mobility
“The overlapping parameters for roaming-sensitive settings are Media Resource Group List, Location, and Network Locale. The overlapping parameters for the Device Mobility-related settings are Calling Search Space (called Device Mobility Calling Search Space at the device pool), AAR Group, and AAR Calling Search Space. Overlapping parameters configured at the phone have higher priority than settings at the home device pool and lower priority than settings at the roaming device pool.”
10. VCS Control routing configuration, user dial brchoi and call gets routed to brchoi@cisco.com
search rule

http://www.manualslib.com/manual/841592/Cisco-Telepresence.html?page=168

http://www.cisco.com/en/US/tech/tk652/tk701/technologies_configuration_guide_chapter09186a00 800eadee.htmlhttp://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/config_guide/X8-2/Cisco-VCS-Basic-Configuration-Control-with-Expressway-Deployment-Guide-X8-2.pdf (Page 17)
“The pre-search transform configuration described in this document is used to standardize destination aliases originating from both H.323 and SIP devices. The following transform modifies the destination alias of all call attempts made to destination aliases which do not contain an ‘@’. The old destination alias has @example.com appended to it. This has the effect of standardizing all called destination aliases into a SIP URI format.”
From VCS and CUCM Deployment guide:
“Thus, a transform is needed to ensure that the dialed number is transformed into a consistent form, in this case to add the domain (vcs.domain) if required.”
11. Configure VG310/VG350 and enable call pickup feature
SCCP gateway
You must check this on a running CUCM. Check CUCM configuration and VG350 gets configurred as SCCP only and then the endpoints can be configured to do a call pick-up.
Check CUCM configuration and VG350 gets configurred as SCCP only and then the endpoints can be configured to do a call pick-up.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmsys/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100/CUCM_BK_SE5FCFB6_00_cucm-system-guide-100_chapter_0100110.html#CUCM_RF_V83D221A_00
12. Intracluster URI dialing configuration

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.html#CUCM_TK_SBE2D597_00

URI Dialing within the same cluster, follow these steps:
Step 1: Configure the URIs to the users
Step 2: Associate the directory URIs to directory numbers
Step 3: Assign the default directory URI (Configure the directory URI partition and calling search space)
Step 4: Configure the SIP profile in your network. (Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network. Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.)
13. Enabling video desktop sharing between CUCM video endpoint and Cisco VCS video endpoint.
Use BFCP
https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=88574&backBtn=true
BRKCOL-2540 – Video call control and management migration to CUCM (2015 Cancun) – 90 Mins

22222

 

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_6_1/ccmcfg/bccm-861-cm/b06siprf.html
Table 76-1 SIP Profile Configuration Settings
Allow Presentation Sharing using BFCP
If the box is checked, Cisco Unified Communications Manager is configured to allow supported SIP endpoints to use the Binary Floor Control Protocol to enable presentation sharing.
The use of BFCP creates an additional media stream in addition to the existing audio and video streams. This additional stream is used to stream a presentation, such as a PowerPoint presentation from someone’s laptop, into a SIP videophone.
If the box is unchecked, Cisco Unified Communications Manager rejects BFCP offers from devices associated with the SIP profile by setting the BFCP application line and associated media line ports to 0 in the answering SDP message. This is the default behavior.
Note BFCP is only supported on SIP networks. BFCP must be enabled on all SIP trunks, lines, and endpoints for presentation sharing to work. BFCP is not supported if the SIP line or SIP trunk uses MTP, RSVP, TRP or Transcoder.
For more information on BFCP, refer to the Cisco Unified Communications Manager System Guide.
14. “Src-port=”25723″ Detail=”Incorrect authentication credential for user”” error
The Expressway-C Traversal Client username/password do not match the Expressway-E Traversal Server username/password.

https://www.ciscolive.com/online/connect/sessionDetail.ww?SESSION_ID=89093&backBtn=true

The Expressway-C is a Client and the Expressway-E is the server. They have client to server relationship. Expressway-C is a Traversal Client.

1111
http://www.cisco.com/c/en/us/support/docs/unified-communications/expressway-series/117811-configure-vcs-00.html

Traversal Zone

When the peer address is configured as an IP address or the peer address does not match the Common Name (CN), you see this in the logs:

Event=”Outbound TLS Negotiation Error” Service=”SIP” Src-ip=”10.48.80.161″
Src-port=”25697″ Dst-ip=”10.48.36.171″ Dst-port=”7001″ Detail=”Peer’s TLS
certificate identity was unacceptable” Protocol=”TLS” Common-name=”10.48.36.171″
When the password is incorrect, you see this in the Expressway-E logs:

Module=”network.ldap” Level=”INFO”: Detail=”Authentication credential found in
directory for identity: traversal”

Module=”developer.nomodule” Level=”WARN” CodeLocation=”ppcmains/sip/sipproxy/
SipProxyAuthentication.cpp(686)” Method=”SipProxyAuthentication::
checkDigestSAResponse” Thread=”0x7f2485cb0700″: calculated response does not
match supplied response, calculatedResponse=769c8f488f71eebdf28b61ab1dc9f5e9,
response=319a0bb365decf98c1bb7b3ce350f6ec

Event=”Authentication Failed” Service=”SIP” Src-ip=”10.48.80.161″
Src-port=”25723″ Detail=”Incorrect authentication credential for user”
Protocol=”TLS” Method=”OPTIONS” Level=”1″
15. An effective backup method to access TEHO destinations in case the call limit triggers
LRG

https://books.google.com.au/books?id=80iuCwAAQBAJ&pg=PT151&lpg=PT151&dq=device+pool+used+to+globalize+dial+plan&source=bl&ots=uyI8e5Jt4L&sig=hZ-hEJ-XCgJT7pZHCrGR89Q5TR4&hl=en&sa=X&ved=0ahUKEwi0n-33iqLNAhXFIqYKHWdaAVEQ6AEIQjAF#v=onepage&q=device%20pool%20used%20to%20globalize%20dial%20plan&f=false
“If TEHO is configured, the appropriate TEHO Gateway is used for the PSTN call. The TEHO route list can include the Default Local Route Group setting as a backup path. In this cas, if the primary (TEHO) path is not available, the gateway taht is referenced by the local route group of the applicable device pool will be used for the backup path. If the device pool selection is not static, but Cisco Unified device mobility is used, the gateway of the roaming site will be used as a backup for the TEHO path. …”
16. Functionalities of subzones in a Cisco VCS deployment
Apply registration, authentication, and media encryption policies
Manage bandwidth to restrict standard definition endpoints from using more than 2 Mb of bandwidth.

http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/admin_guide/Cisco_VCS_Administrator_Guide_X7-2.pdf (Page 127)

http://www.cisco.com/c/dam/en/us/td/docs/telepresence/infrastructure/vcs/admin_guide/Cisco-VCS-Administrator-Guide-X8-7-2.pdf (Page 154)

Bandwidth management
The Local Zone’s subzones are used for bandwidth management. After you have set up your subzones you can apply
bandwidth limits to:
– individual calls between two endpoints within the subzone
– individual calls between an endpoint within the subzone and another endpoint outside of the subzone
– the total of calls to or from endpoints within the subzone

For full details of how to create and configure subzones, and apply bandwidth limitations to subzones including the
Default Subzone and Traversal Subzone, see the Bandwidth control section.

Registration, authentication and media encryption policies
In addition to bandwidth management, subzones are also used to control the VCS’s registration, authentication and
media encryption policies.
17. Enabling SAF Call Control Discovery
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_2/ccmfeat/fsgd-802- cm/fscallcontroldiscovery.pdf
1. the SIP or H.323 trunk
2. hosted DN patterns
3. Hosted DN groups
18. Cisco VCS Expressway traversal call licenses
http://www.cisco.com/c/en/us/support/docs/unified-communications/telepresence-video-communication-server-vcs/118872-technote-vcs-00.html
According to the document VCS, Gatekeepers and Border Controllers. SIP Trunk is treated as a device by Cisco, but it is not a real device, so not used for licensing.
19. Devices or applications support call preservation
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/CUCM_BK_CD2F83FA _00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system- guide_chapter_01011.html#CUCM_RF_C98194B0_00

The following devices and applications support call preservation. If both parties connect through one of the following devices, Cisco Unified Communications Manager maintains call preservation:
Cisco Unified IP Phones
SIP trunks
Software conference bridge
Software MTP
Hardware conference bridge (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Transcoder (Cisco Catalyst 6000 8 Port Voice E1/T1 and Services Module, Cisco Catalyst 4000 Access Gateway Module)
Non-IOS MGCP gateways (Catalyst 6000 24 Port FXS Analog Interface Module, Cisco DT24+, Cisco DE30+, Cisco VG200)
Cisco IOS H.323 gateways (such as Cisco 2800 series, Cisco 3800 series)
Cisco IOS MGCP Gateways (Cisco VG200, Catalyst 4000 Access Gateway Module, Cisco 2620, Cisco 3620, Cisco 3640, Cisco 3660, Cisco 3810)
Cisco VG248 Analog Phone Gateway

The following devices and applications do not support call preservation:
Annunciator
H.323 endpoints such as NetMeeting or third-party H.323 endpoints
CTI applications
TAPI applications
JTAPI applications
Call Preservation Scenarios
20. Global Dial Plan Replication prevent the local cluster from routing VIP number 6666666666 to the remote cluster.
Create a block learned pattern.
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/10_0_1/ccmfeat/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100/CUCM_BK_F3AC1C0F_00_cucm-features-services-guide-100_chapter_011101.html#CUCM_RF_L56BD6F6_00
Learned pattern can be configured but there is no mentioning of transformation pattern configuration. “Create a block learned pattern” can be used to prevent Global Dial Plan Replication within local cluster.
21. URI calling within the same cluster configuration

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmsys/CUCM_BK_CD2F83FA_00_cucm-system-guide-90/CUCM_BK_CD2F83FA_00_system-guide_chapter_0101111.html#CUCM_TK_SBE2D597_00

URI Dialing within the same cluster, follow these steps:
Step 1: Configure the URIs to the users
Step 2: Associate the directory URIs to directory numbers
Step 3: Assign the default directory URI (Configure the directory URI partition and calling search space)
Step 4: Configure the SIP profile in your network. (Configure a setting for the Dial String Interpretation drop-down list box and apply the setting for all the SIP profiles in your network. Check the Use Fully Qualified Domain Name in SIP Requests check box for all the SIP profiles in your network.)
22. 2 steps must you take when implementing TEHO in your environment
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/7x/uc7_0/dialplan.html
Implement local failover
Implement centralized failover
23. Globalization dialing functions enhancement since CUCM 7.X and later
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/9x/uc9x/dialplan.html (benefits of new design approach)
AAR
CER
TEHO
24. 2 commands verify Cisco IP Phone registration
show ephone registered
show sip-ua status registrar

http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/troubleshooting/guide/ts_phreg.html (see the steps)

Step 3 show sip-ua status registrar

Use this command to display all the SIP endpoints currently registered with the contact address.

Router# show sip-ua status registrar

Line destination expires(sec) contact
============ =============== ============ ===============
91021 192.168.0.3 227 192.168.0.3
91011 192.168.0.2 176 192.168.0.2
95021 10.2.161.50 419 10.2.161.50
95012 10.2.161.50 419 10.2.161.50
95011 10.2.161.50 420 10.2.161.50
95500 10.2.161.50 420 10.2.161.50
94011 10.2.161.40 128 10.2.161.40
94500 10.2.161.40 129 10.2.161.40

I have cross checked with my Voice Gateway and found that ‘show ephone registered’ and ‘show sip-ua status registrar’.

Router#show ephone registered

ephone-1[0] Mac:6C30.4D57.8CD5 TCP socket:[1] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.34 * 7962 keepalive 4929 music 0 1:101 CM Fallback
sp1:01800008584 sp2:01800654112 sp3:00362456600

ephone-2[1] Mac:555D.0608.45B6 TCP socket:[-1] activeLine:0 whisperLine:0 UNREGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.15 * 6921 keepalive 7 music 0

ephone-3[2] Mac:448D.0407.6BE9 TCP socket:[4] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.124 * 6921 keepalive 4938 music 0 1:103 CM Fallback

ephone-4[3] Mac:544D.0907.532C TCP socket:[12] activeLine:0 whisperLine:0 REGISTERED
mediaActive:0 whisper_mediaActive:0 startMedia:0 offhook:0 ringing:0 reset:0 reset_sent:0 debug:0
IP:10.30.11.25 * 6921 keepalive 4931 music 0 1:2 CM Fallback
25. Enalbe presence and extension mobility to branch office phones during a WAN failure. Cisco Unified Communications Manager Express in SRST mode

http://www.ciscopress.com/articles/article.asp?p=1744068&seqNum=4
Cisco Unified SRST does not support enhanced features, such as Presence or Cisco Extension Mobility. Message Waiting Indicator (MWI) is also not supported in fallback mode.

 

26. Configured a Cisco EX60 to register with a Cisco VCS Control, but phone is not registering with VCS C. What’s missing in the configuration.
http://www.cisco.com/en/US/docs/telepresence/infrastructure/vcs/config_guide/X8-1/Cisco-VCS-Basic-Configuration-Single-VCS-Control-Deployment-Guide-X8-1.pdf

EX60 (uses H.323 and SIP protocol)
H.323 ID user.two.mxp@example.com
H.323 E.164 7654321
Gatekeeper IP Address vcsc.internal-domain.net
SIP URI user.two.mxp@example.com
SIP Proxy1 vcsc.internal-domain.net
EX90 (uses SIP protocol)
SIP URI user.one.ex90@example.com
SIP Proxy1 vcsc.internal-domain.net
27. Your company’s internal number is 4 digit dialing, how to present this as 10-digit number to external clients?
Use “calling party transformation pattern”
https://supportforums.cisco.com/discussion/9848251/external-phone-mask-vs-calling-party-transformation-mask
“An advantage of using Calling Party Transformation Mask is that it allows you to change the Calling party number for a bunch of phones easily. Lets say you have a 100 phones that you need to change the 10 digit number. Rather than going to each phone and change the setting individually, you can do it at the Calling Party tranformation mask.”
“Another advantage is that if you want to change Calling Party number that gets displayed to external users, you can modify that easily with the transformation masks. It also gives you the flexibility of sending different calling party numbers to differnt destinations. For example, for local calls you can dislay the 7 digit number; for long distance you can display 10 digits and for international you can display country code +10 digits.”
28. Default region configurable items on CUCM?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/8_0_1/ccmcfg/bccm-801- cm/b02regio.html#wp1077135
Audio Codec
Video Call Bandwidth
Link Loss Type
29. During Intercluster URI dialing, an error message “Local cluster cannot connect to the ILS network” comes up, what could be possible issues?
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/admin/9_0_1/ccmfeat/CUCM_BK_CEF0C471_00_cucm-features-services-guide-90/CUCM_BK_CEF0C471_00_cucm-features-and-services-guide_chapter_011111.pdf (Page. 8)
The Tomcat certificates do not match.
The ILS authentication password does not match.
One cluster is using TLS certificate, and the other is using Password.
30. 2 technologies not utilising MTP.
http://www.cisco.com/c/en/us/td/docs/ios/voice/cube/configuration/guide/vb_8241.html

DTMF inband RTP-NTE (rfc2833)

SIP Delay Offer

Note 1: H.323 fast start:
https://learningnetwork.cisco.com/thread/65786
DTMF inband RTP-NTE (rfc2833) requires MTP only in CM 4.0, 5 and in later versions of CUCM, and lMTP requirement was removed when supporting RFC 2833 DTMF)

Note 2: If both endpoints support NTE, then no MTP is required. http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/8x/uc8x/media.html#wp1046314

Chasing Packets in GNS3 & Production Environment, Part 2: IOS Embedded Packet Capture & tee off to a TFTF server

aaa2

IOS Embedded Packet Capture Configuration in a nutshell:

r1#monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

r1#monitor capture point ip cef E0_0 e0/0 both

r1#monitor capture point associate E0_0 PAKCETBUFFER

r1#monitor capture point start E0_0

 

Generate some traffic: example showing ICMP traffic generation

aaa9

Generate some RTP traffic: example showing use of Cisco IP communicator in this lab

aaa11.png

 

r1#monitor capture point stop E0_0

r1#monitor capture buffer PAKCETBUFFER export tftp://172.168.10.10/mycapture.pcap

 

***You must specify the name of the file, otherwise the teeing off to TFTP server will not work!!!

 

aaa8

 

Example of ICMP traffic packet capture:

aaa10.png

 

Example of RTP traffic packet capture.

aaa7

 

 

=======================================================================

Actual configuration:

r1#monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

 

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 0

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Inactive

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

 

r1#mon cap point ip cef E0_0 e0/0 both

*Apr  5 07:30:22.526: %BUFCAP-6-CREATE: Capture Point E0_0 created.

 

r1#show mon cap point all

Status Information for Capture Point E0_0

IPv4 CEF

Switch Path: IPv4 CEF            , Capture Buffer: None

Status : Inactive

 

Configuration:

monitor capture point ip cef E0_0 Ethernet0/0.100 both

 

r1#mon cap point associate E0_0 PAKCETBUFFER

 

r1#show mon cap point all

Status Information for Capture Point E0_0

IPv4 CEF

Switch Path: IPv4 CEF            , Capture Buffer: PAKCETBUFFER

Status : Inactive

 

Configuration:

monitor capture point ip cef E0_0 Ethernet0/0.100 both

 

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 0

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Inactive

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

 

 

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 0

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Active

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 3

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Active

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 4

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Active

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 657

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Active

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

r1#mon cap point stop E0_0

r1#mon cap point stop E0_0

*Apr  5 07:34:11.582: %BUFCAP-6-DISABLE: Capture Point E0_0 disabled.

r1#show mon cap buffer PAKCETBUFFER parameters

Capture buffer PAKCETBUFFER (linear buffer)

Buffer Size : 2097152 bytes, Max Element Size : 128 bytes, Packets : 657

Allow-nth-pak : 0, Duration : 0 (seconds), Max packets : 0, pps : 0

Associated Capture Points:

Name : E0_0, Status : Inactive

Configuration:

monitor capture buffer PAKCETBUFFER size 2048 max-size 128 linear

monitor capture point associate E0_0 PAKCETBUFFER

 

 

r1#show monitor capture buffer PAKCETBUFFER dump

07:33:16.228 UTC Apr 5 2016 : IPv4 LES CEF    : Et0/0.100 None

 

F4E2C230: AABBCC00 0100000C 2978156D 81000064  *;L…..)x.m…d

F4E2C240: 08004560 0034EE83 40007F06 1959C0A8  ..E`.4n.@….Y@(

F4E2C250: 640B8EC8 400B0570 07D079AA FEBF61F3  d..H@..p.Py*~?as

F4E2C260: 050A5018 FAC0BFD0 00000400 00001100  ..P.z@?P……..

F4E2C270: 00000000 000000                      …….

… Content omitted for brevity

 

 

r1#monitor capture buffer PAKCETBUFFER export ?

disk0:  Location to dump buffer

disk1:  Location to dump buffer

ftp:    Location to dump buffer

http:   Location to dump buffer

https:  Location to dump buffer

pram:   Location to dump buffer

rcp:    Location to dump buffer

scp:    Location to dump buffer

snmp:   Location to dump buffer

tftp:   Location to dump buffer

unix:   Location to dump buffer

 

r1#monitor capture buffer PAKCETBUFFER export tftp://172.168.10.10/mycapture.pcap

!

***You must specify the name of the file, otherwise the teeing off to TFTP server will not work!!!

 

Chasing Packets in GNS3 & Production Environment, Part 1: Capturing packets using built-in Live Wireshark Capture in GNS3 1.4.4

Why do you want to do this lab?

You can capture any interesting packets and analyse for your learning purpose, analyzing packet captures can give you the real inside of how the packets are working on the devices and on different segments of the network. Simply reading the books and learn about how packets work behind the scenes is a little like trying to learn something as if you are three wise monkeys (see no evil, hear no evil, speak no evil).

On the real production, you can use other methods to capture interesting packets. Some examples are IOS Embedded Packet capture and tee off the configuration to a TFTP server, use a sniffer using spanning port or remote spanning port. Also, use more advanced method of Cisco NAM (Network Analyzer).

In this part, I will quickly show you how to whiz up a simple lab and capture some packets on GNS3 and Wireshark live capture within, GNS3. In the next section, I will demonstrate IOS Embedded Packet capture and teeing off to a TFTP server. Lastly, I will demonstrate packet capturing using spanning port and remote span.

Prerequisite 1: GNS3 1.4.4 pre-installed on Windows PC/laptop

Prerequisite 2: IOU VM ova deployed and integrated with GNS3

Prerequisite 3: Familiar with VMware workstation and Windows loopback configuration

 

Topology:

aaa1

Step 1: Add devices as below and make all connections. When you add the devices, your GNS3 topology will look like this. Remember to use dummy switches to make connection between your virtual machines and your host PC loopback to your IOU switches.

aaa2.png

Step 2: Configure your routers and switches similar to the configuration found in  the attached zip file.

r1

r2

sw1

sw2

 

Step 3: Capture packets using various link positions

aaa3

aaa4

If you run into the following error, you will have to go to GNS3 setting and update the path of Wireshark.

aaa5

=> Error: SW3: Could not start the packet capture reader: [WinError 2] The system cannot find the file specified: None

Changing path in GSN3 preferences:

C:\Program Files\Wireshark\wireshark.exe” ==> C:\Program Files (x86)\Wireshark\wireshark.exe

 

Step 4: Wireshark will open automatically and start capturing all the traffic on the link you have selected.

e.g.) TCP/IP packet capture example

aaa6.png

e.g.) Voice packet capture using soft phones (On virtual machines) between two work stations and CUCM.

aaa7

Now you can set up any server and clients and study how TCP/IP, UDP work behind the scenes. Jump straight in and try to enjoy your study!

 

Note: This lab can be completed on a single PC, Save Electricity, save Money, save Time, SAVE THE PLANET.

 

 

 

Notes on Cisco QoS: Clearing the fog – Part 4. Modular QoS Lab

Lab topology:

Module QoS 2

How this lab can be configured in GNS3 on a single PC.

  • SW1 and SW2 is the local GNS3 switches, merely serving as a connector between PC1 and HTTP Server respectively. These dummy switches must be used while connecting virtual machines to GNS3 devices.

Module QoS 1

Step 1: Configure R1 and R2 to allow communication between the networks.

R1 base configuration:

hostname R1

interface FastEthernet0/0
ip address 192.168.30.254 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0
ip address 1.1.1.1 255.255.255.0
clock rate 2000000
!
router eigrp 1
network 1.0.0.0
network 192.168.30.0
auto-summary

==============================================

R2 base configuration:

hostname R2

interface FastEthernet0/0
ip address 192.168.40.254 255.255.255.0
duplex auto
speed auto

router eigrp 1
network 1.0.0.0
network 192.168.40.0
auto-summary

==============================================

Step 2: Configure R1 with Access List, class-map and policy-map

access-list 200 permit icmp host 192.168.30.30 host 192.168.40.40 echo
access-list 200 permit icmp host 192.168.30.30 host 192.168.40.40 echo-reply
access-list 100 permit tcp any any eq www

class-map match-all WEB_TRAFFIC
match access-group 100
class-map match-all ICMP_TRAFFIC
match access-group 200

policy-map MODULAR
class ICMP_TRAFFIC
bandwidth 256
class WEB_TRAFFIC
bandwidth 128
class class-default

Step 3: Apply policy map to output queue of Serial 0/0

!Apply Service-policy to output interface s0/0

interface Serial0/0
ip address 1.1.1.1 255.255.255.0
clock rate 2000000
 service-policy output MODULAR

==============================================

Step 4: Run quick check on the configuration

R1#show class-map
Class Map match-all WEB_TRAFFIC (id 1)
Match access-group  100

Class Map match-any class-default (id 0)
Match any

Class Map match-all ICMP_TRAFFIC (id 2)
Match access-group  200

R1#show policy-map
Policy Map CCIE
Class ICMP_TR
Bandwidth 128 (kbps) Max Threshold 64 (packets)
Class WEB_TR
Bandwidth 64 (kbps) Max Threshold 64 (packets)
Class class-default

==============================================

Before any ping or http traffic is sent across the WAN link

R1#show policy-map interface s0/0
Serial0/0

Service-policy output: MODULAR

Class-map: ICMP_TRAFFIC (match-all)
    0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: access-group 112
Queueing
Output Queue: Conversation 265
Bandwidth 128 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: WEB_TRAFFIC (match-all)
      0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: access-group 120
Queueing
Output Queue: Conversation 266
Bandwidth 64 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: class-default (match-any)
697 packets, 46091 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: any

==============================================

Step 5: Generate ICMP traffic by pining the server from the client PC

To generate ICMP traffic, from the client PC (192.168.30.30) ping http server at 192.168.40.40.
ICMP pinging

‘show policy-map interface s0/0’ after 8 ping messages have been sent from 192.168.30.30 (client) to 192.168.40.40 (Server)

R1#show policy-map interface s0/0
Serial0/0

Service-policy output: MODULAR

Class-map: ICMP_TRAFFIC (match-all)
8 packets, 512 bytes <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
5 minute offered rate 0 bps, drop rate 0 bps
Match: access-group 112
Queueing
Output Queue: Conversation 265
Bandwidth 128 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: WEB_TRAFFIC (match-all)
0 packets, 0 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: access-group 120
Queueing
Output Queue: Conversation 266
Bandwidth 64 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: class-default (match-any)
766 packets, 50456 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: any

==============================================

Step 6: Access web page of http server from the client PC

To generate some http traffic, access http://192.168.40.40/ from the client PC to HTTP Server.
Access IIS

==============================================

show policy-map interface serial0/0 after generating http traffic

R1#show policy-map interface s0/0
Serial0/0

Service-policy output: MODULAR

Class-map: ICMP_TRAFFIC (match-all)
    12 packets, 768 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: access-group 112
Queueing
Output Queue: Conversation 265
Bandwidth 128 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: WEB_TRAFFIC (match-all)
13 packets, 2539 bytes <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<
5 minute offered rate 0 bps, drop rate 0 bps
Match: access-group 120
Queueing
Output Queue: Conversation 266
Bandwidth 64 (kbps)Max Threshold 64 (packets)
(pkts matched/bytes matched) 0/0
(depth/total drops/no-buffer drops) 0/0/0

Class-map: class-default (match-any)
878 packets, 57842 bytes
5 minute offered rate 0 bps, drop rate 0 bps
Match: any

 

==============================================

R1#show ip route
Codes: C – connected, S – static, R – RIP, M – mobile, B – BGP
D – EIGRP, EX – EIGRP external, O – OSPF, IA – OSPF inter area
N1 – OSPF NSSA external type 1, N2 – OSPF NSSA external type 2
E1 – OSPF external type 1, E2 – OSPF external type 2
i – IS-IS, su – IS-IS summary, L1 – IS-IS level-1, L2 – IS-IS level-2
ia – IS-IS inter area, * – candidate default, U – per-user static route
o – ODR, P – periodic downloaded static route

Gateway of last resort is not set

1.0.0.0/8 is variably subnetted, 2 subnets, 2 masks
C       1.1.1.0/24 is directly connected, Serial0/0
D       1.0.0.0/8 is a summary, 00:59:46, Null0
C    192.168.30.0/24 is directly connected, FastEthernet0/0
D    192.168.40.0/24 [90/2195456] via 1.1.1.2, 00:59:41, Serial0/0

 

R2#show ip route
Codes: C – connected, S – static, R – RIP, M – mobile, B – BGP
D – EIGRP, EX – EIGRP external, O – OSPF, IA – OSPF inter area
N1 – OSPF NSSA external type 1, N2 – OSPF NSSA external type 2
E1 – OSPF external type 1, E2 – OSPF external type 2
i – IS-IS, su – IS-IS summary, L1 – IS-IS level-1, L2 – IS-IS level-2
ia – IS-IS inter area, * – candidate default, U – per-user static route
o – ODR, P – periodic downloaded static route

Gateway of last resort is not set

1.0.0.0/8 is variably subnetted, 2 subnets, 2 masks
C       1.1.1.0/24 is directly connected, Serial0/0
D       1.0.0.0/8 is a summary, 00:04:16, Null0
D    192.168.30.0/24 [90/2195456] via 1.1.1.1, 00:04:11, Serial0/0
C    192.168.40.0/24 is directly connected, FastEthernet0/0

 

All this lab was done on a laptop, go easy on the environment. 🙂

On a single PC

Notes on Cisco QoS: Clearing the fog – Part 2. Quality issues

Quality of Service

QOS = Method of giving priority to some specific traffic as moving over the network.

The basic aim of QoS is to have a consistent and predictable performance on your network.

 

1 qos intro

General characteristics of today’s Converged Network:

  • Small voice packet compete with bursty data packets, many different applications are using network as services
  • Critical traffic must get priority over less critical traffic, without QoS, default behavior is First In First Out (FIFO)
  • Voice and video traffics are time-sensitive
  • Outages are not acceptable

 

Converged Network Quality issues:

  • Lack of Bandwidth
  • Packet Loss
  • Delay
  • Jitter

 

Bandwidth

2 Bandwidth Measure.png

  • Maximum available bandwidth is the slowest link on the traffic paths
  • On the same physical links (traffic paths), multiple flows compete for the same bandwidth, multiple applications sharing the same bandwidth
  • Lack of bandwidth causes performance degradation on network applications

 

 

Packet Loss

3 Tail Drop due to Queue Congestion

Packet loss due to Tail Drop: Queue only can so much packets and once it is full and more packets arrive at the tail end of the queue before the queue is emptied (due to link congestion etc.), the packets will be dropped, and this behavior is called ‘Tail Drop’. If the tail drop occurs to the time sensitive traffics such as voice and video, the effects are immediately felt by the users on the flow. If this happens to data traffic, it may interrupt file transfer and corrupt the file.

 

 

Delay

4 Types of Delay

  • Processing Delay – time taken by router to process packets from an input interface and put them into the output queue of output interface
  • Queuing Delay – time a packet resides in the output queue of a router
  • Serialization Delay – time taken to place bits on the wire
  • Propagation Delay – time taken for packets to cross links from one end to the other end

 

 

Jitter

5 Jitter

  • Packets from a source will reach a destination with different delay times
  • Congestion on the network will cause jitter
  • Congestion can occur at a router interface/Service Provider network if the circuits are not properly provisioned

 

CUBE High Availability (HA) Using HSRP Configuration with port-channel twist

Starting with Cisco Gen2 router platforms, CUBE can provide the HSRP (Hot Standby Routing Protocol). That is you need two CUBE routers to confgure this setup. HSRP basically works on Active and Standby mode between two routers by monitoring both the inside and outside interfaces, if Active side goes down, then the Standby device becomes active and takes over the responsibilities of the Active router.

In CUBE HSRP Active/Standby pair scenario, the two CUBE routers keep exchange communications over the same virtual IP address. This setup will support media preservation over an HSRP switchover of SIP to SIP calls, but not the call signaling. Call signaling preservation is supported from IOS 15.2.3T.

Requirements:
1. Two identical ISR G2 routers with the correct IOS and license
*Cisco 2951 (x 2), IOS = c2951-universalk9-mz.SPA.154-3.M1, license =SL-29-UC-K9
2. Identiacal CUBE configuration
3. SIP-to-SIP call flows
Configuration:
1. Enable CUBE and CUBE Redundancy

Enable CUBE on CUBE01 and CUBE02:
voice service voip
mode border-element
allow-connections sip to sip

Enable CUBE redundancy and call checkpointing on both CUBES
voice service voip
redundancy
2. Enable HSRP

Enable router redundancy schemes on both routers, where:
scheme – redundancy state tracking scheme
standby – enable standby (HSRP) state tracking scheme
SB – the HSRP standby group name

redundancy inter-device
scheme standby SB

3. Configure HSRP Communication Transport

Configure the HSRP Inter-Device Communication Transport as follows:

Active Configuration:
ipc zone default <<< For Inter-Device Communication Protocol (IPC)
association 1 <<< Associates between two devices
no shutdown <<< Enables associations
protocol sctp <<< Stream Control Transmission Protocol (SCTP) for communication language
local-port 5000 <<< Defines the local SCTP port number
local-ip 10.10.24.14 <<< Defines the local router’s IP address
remote-port 5000 <<< Defines the remote SCTP port number
remote-ip 10.10.24.13 <<< Defines the remote router’s IP address

Standby Configuration:
ipc zone default
association 1
no shutdown
protocol sctp
local-port 5000
local-ip 10.10.24.13
remote-port 5000
remote-ip 10.10.24.14

4. Configure HSRP on the Interfaces

Configure the HSRP Inter-Device Communication Transport as follows:

Active Configuration

interface Port-channel1
description CUBE01 interface
ip address 10.10.10.11 255.255.255.0
standby delay minimum 30 reload 60 <<< Avoids race condition to establish contact between Active and Standby
standby version 2
standby 0 ip 10.10.10.1
standby 0 priority 50
standby 0 preempt
standby 0 name SB

interface GigabitEthernet0/0
no ip address
duplex full
speed 1000
channel-group 1

interface GigabitEthernet0/1
no ip address
duplex full
speed 1000
channel-group 1
Standby Configuration:

interface Port-channel1
description CUBE02 interface
ip address 10.10.10.12 255.255.255.0
standby delay minimum 30 reload 60
standby version 2
standby 0 ip 10.10.10.1
standby 0 priority 50
standby 0 preempt
standby 0 name SB

interface GigabitEthernet0/0
no ip address
duplex full
speed 1000
channel-group 1

interface GigabitEthernet0/1
no ip address
duplex full
speed 1000
channel-group 1
5. Configure the HSRP Timers

CUBE01(config-if)#standby 0 timers 2 msec 40 <<< configures failover and hold timers

CUBE02(config-if)#standby 0 timers 2 msec 40
6. Configure the Media Inactivity Timer

Enables the Active/Standby router pair to monitor and disconnect calls if no Real-Time Protocol (RTP) packets are received within a configurable time period. Default value is 28 seconds.

ip rtcp report interval 3000
gateway
media-inactivity-criteria all
timer receive-rtp 86400
timer receive-rtcp 5
7. Configure SIP Binding to HSRP Address

voice service voip
mode border-element license capacity 125
allow-connections sip to sip
redundancy
sip
bind control source-interface Port-channel1
bind media source-interface Port-channel1
asserted-id pai
asymmetric payload full
midcall-signaling passthru
privacy-policy passthru
sip-profiles 100
8. Reload the Routers

Active Router
CUBE01#show redundancy inter-device
Redundancy inter-device state: RF_INTERDEV_STATE_ACT
Scheme: Standby
Groupname: b2bha Group State: Active
Peer present: RF_INTERDEV_PEER_COMM
Security: Not configured

Standby Router
CUBE02#show redundancy inter-device
Redundancy inter-device state: RF_INTERDEV_STATE_STDBY
Scheme: Standby
Groupname: b2bha Group State: Standby
Peer present: RF_INTERDEV_PEER_COMM
Security: Not configured
9. Point Attached Softswitches to the CUBE HSRP Virtual Address
On CUCM, this is configured on the SIP Trunk configuration under Device > Trunk.

SIP Trunk

 

**********************************************
Useful commands for verification and troubleshooting:
show redundancy inter-device
show redundancy states
show standby brief
show standby
show voice high-availability summary
show voice high-availability summary | include media
show voip rtp connection
show sip-ua status
show sip-ua statistics
debug standby

show process cpu history
show process cpu sorted

***********************************************