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Archive for the ‘Gateways’ Category

IP Traffic Exporting CUBE

November 28, 2014 Leave a comment

Hi All

Ever wanted to capture all traffic for a voice call on an interface on a CUBE

Below are the commands required to capture the info, it can then be opened in Wireshark for troubleshooting.

ip traffic-export profile <name> mode capture
bidirectional

interface <interface>
ip traffic-export apply <name> size 20000000

traffic-export interface <interface> clear
traffic-export interface <interface> start

Make test call

traffic-export interface <interface> stop
traffic-export interface <interface> copy ftp:

Enjoy

TheVoiceMan

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Categories: Gateways Tags: , ,

Clear Hung/Stale calls on CUBE

November 27, 2014 Leave a comment

Hi All

Have implemented several SIP services recently from all carriers and have found that sometimes the calls either don’t end correctly or some SIP call legs drop off…..

When this occurs the CUBE does not correctly remove these call legs and we end up with hung calls or stale calls on the CUBE.

If you have the “Max Connections” command configured you may find that these hung/stale calls will add to the total therefore hitting the “Max Connections” earlier than expected.

To check this run the following command

sh call active voice summary

Telephony call-legs: 0

SIP call-legs: 4

H323 call-legs: 0

Call agent controlled call-legs: 0

SCCP call-legs: 0

Multicast call-legs: 0

Total call-legs: 4

You can also use the following two commands to view the details of the calls

sh rtp call

show sip calls

Use the “called” number from the show commands given above and use the “clear call voice” command to clear the hung calls towards the ISP network with the Call ID and Cause Code Value of the hung call:

Clear Call Voice command: To clear one or more voice calls detected as inactive because there is no RTP or RTCP activity, use the clear call voice command in EXEC or privileged EXEC mode.

hung-1.png

Enjoy

TheVoiceMan

Categories: Gateways Tags: , , , ,

Viewing CUCM Trace Files and CUBE Debugs

November 25, 2014 Leave a comment

Hi All

Thought I would post this up as I have been using this tool for ages now and assumed that most people know about it.

The TranslatorX tool is an awesome tool for viewing Debugs from a CUBE or loading in Trace files from RTMT for a CUCM issue, it clearly outputs line by line the debugs and traces and you can even load a folder containing multiple files all at the same time.

Try it out, great tool and best of all it is free……

http://translatorx.cisco.com

Enjoy

TheVoiceMan

Categories: CUCM, Gateways Tags: , , ,

Forcing Cisco IP Phones into SRST mode

February 19, 2013 Leave a comment

Hi All

Have you ever wanted to know how to force your remote site Cisco IP Phones into SRST mode without turning off the CUCM servers or shutting down your WAN link. This can be achieved by using a access list on the router/WAN interface

Configure the access list that blocks the following

SIP: 5060 (TCP/UDP),
Secure SIP: 5061 (TCP/UDP),
SCCP: 2000 (TCP)
Secure SCCP: 2443 (TCP)
Standard RTP Ports: 16384-32767 (UDP)

ip access-list extended ACL-VOIP-CONTROL
deny tcp any any eq 5060
deny udp any any eq 5060
deny tcp any any eq 5061
deny udp any any eq 5061
deny tcp any any eq 2000
deny tcp any any eq 2443
deny udp any any range 16384 32767
permit ip any any

Apply the access control list to the WAN interface to which the administrator wishes to block voice traffic.

interface X/X
ip access-group ACL-VOIP-CONTROL in

The phones will now failover into SRST mode

Enjoy

The Voice Man

Categories: CUCM, Gateways

Configuring Outbound Services via Alternate Trunks ISDN & SIP

February 14, 2013 Leave a comment

Hey All

Have had a few questions around how you can determine which trunk a call leaves a customers environment, and how you can force calls through a particular trunk. This is very useful for customers who have multiple services but only have certain number ranges down each service, or if the customer wants to block CLI on one of the trunks and only send certain call out this one and the rest out one of the other trunks.

To set this up a few additional commands and dial-peers need to be configured on the relevant voice gateway to first locate the call with the correct calling number and then ensure it leave the system through the correct trunk to the carrier.

Using the Cisco IOS “answer-address” command this can be achieved

Usage Guidelines

Use the answer-address command to identify the origin (or dial peer) of incoming calls from the IP network. Cisco IOS software identifies the dial peers of a call in one of two ways: by identifying either the interface through which the call is received or the telephone number configured with the answer-address command. In the absence of a configured telephone number, the peer associated with the interface is associated with the incoming call.

For calls that come in from a plain old telephone service (POTS) interface, the answer-address command is not used to select an incoming dial peer. The incoming POTS dial peer is selected on the basis of the port configured for that dial peer.

First create some translation rules and then assign these translation rules to the translation profiles.

These will pre-pend a number for the incoming call to the gateway from CUCM in the below example this is 77 and 88 and remove the 77 and 88 including the zero (0) when it leaves the gateway and goes to the carrier

Note: Try not to use 99 as the prepending digits as this causes issues when you are then trying to strip the digits leaving the gateway

!
voice translation-rule 77
rule 1 // /77/
!
voice translation-rule 88
rule 1 // /88/
!
voice translation-rule 777
rule 1 /^770/ //
!
voice translation-rule 888
rule 1 /^880/ //
!
voice translation-profile PrePend77Incoming
translate called 77
!
voice translation-profile PrePend88Incoming
translate called 88
!
voice translation-profile Remove77Outgoing
translate called 777
!
voice translation-profile Remove88Outgoing
translate called 888
!

The below dial peers capture the calls based on their mask when leaving the CUCM system and matches on one of the below dial peers which in turns forces the calls out the relevant Trunks

dial-peer voice 77 voip
description === Outgoing Calls ALLOW CLI ===
translation-profile incoming PrePend77Incoming
answer-address 024321….
dtmf-relay rtp-nte h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 88 voip
description === Outgoing Calls BLOCK CLI ===
translation-profile incoming PrePend88Incoming
answer-address 029876….
dtmf-relay rtp-nte h245-alphanumeric
codec g711ulaw
no vad
!
dial-peer voice 1 pots
description === Outgoing Calls BLOCK CLI ===
translation-profile outgoing Remove88Outgoing
destination-pattern 88T
progress_ind setup enable 3
progress_ind progress enable 8
port 0/0/0:15
!
dial-peer voice 2 pots
description === Outgoing Calls ALLOW CLI ===
translation-profile outgoing Remove77Outgoing
destination-pattern 77T
progress_ind setup enable 3
progress_ind progress enable 8
port 0/0/1:15
!

Below is also a default dial-peer which has the standard destination-pattern 0T which captures any calls without a mask and sends them out the defined default trunk.

dial-peer voice 3 pots
description ==== Outgoing Calls Default ====
destination-pattern 0T
progress_ind setup enable 3
progress_ind progress enable 8
port 0/0/2:15
!
!

Enjoy

The Voice Man

Categories: CUCM, Gateways

Configuring and Troubleshooting PVDM3 Modules

February 13, 2013 Leave a comment

Hi All

Found a great link to a Cisco document which explains quite well how to confirm and troubleshoot DSP related issues for PVDM3 modules

http://www.cisco.com/en/US/docs/routers/access/1900/software/configuration/guide/pvdm3_config.pdf

Enjoy

The Voice Man

Categories: Gateways

CUBE Configuration Example

May 14, 2012 Leave a comment

Hi All

Great link to a sample CUBE configuration from Cisco for integrating with CUCM

Check it out

CUBE Configuration Example

Cheers

The Voice Man

Categories: CUCM, Gateways Tags: , , ,