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CCIE DataCenter

March 27, 2014

I started working on CCIE DataCenter Blue print to get more hands on experience on the DC technologies.

Fax over a SIP Trunk through a CUBE router

May 19, 2011

must be using G711 codec.
– Configure cube
– configure dial peer
– configure sip trunk
– configure route group, route list, route patter
– create a partition and assign it to the route pattern

voice service voip
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol pass-through g711ulaw
h323
sip
early-offer forced
midcall-signaling passthru
!
!
voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g729r8

dial-peer voice 120 voip
description ** OUTBOUND FAX @ G7.11 **
translation-profile outgoing STRIP+1
destination-pattern 91[2-9]..[2-9]……$
voice-class codec 2
voice-class sip early-offer forced
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte sip-notify
no vad
!

voice translation-rule 1
rule 1 /^91/ /+1/
!
!
voice translation-profile STRIP+1
translate called 1

sip-ua
set pstn-cause 1 sip-status 604
set pstn-cause 102 sip-status 503
retry invite 2
retry bye 2
retry cancel 2
retry options 1
sip-server ipv4:XX.XX.XX.XX

Cisco Unified Communication NIC Teaming for Fault Tolerance

May 5, 2011

The NIC teaming feature allows a server to be connected to the Ethernet via two NICs and, therefore, two cables. NIC teaming prevents network downtime by transferring the workload from the failed port to the working port. NIC teaming cannot be used for load balancing or increasing the interface speed.

Hewlett-Packard server platforms with dual Ethernet network interface cards can support NIC teaming for Network Fault Tolerance with Cisco Unified CM 5.0(1) or later releases.

IBM server platforms with dual Ethernet network interface cards can support NIC teaming for Network Fault Tolerance with Cisco Unified CM 6.1(2) and later releases. “

And the command to do that as per the CUCM OS CLI Reference Guide is:

set network failover

This command enables and disables Network Fault Tolerance on the Media Convergence Server network interface card.

Command Syntax

failover {enable | disable}

Parameters

enable enables Network Fault Tolerance.

disable disables Network Fault Tolerance.

Requirements

Command privilege level: 1

Allowed during upgrade: No “

MOH file conversion to .g729/wb/alaw/ulaw

May 5, 2011

The steps to convert prompts in Linux-based Cisco Unified Communications Manager follow:

1. Log into Cisco Unified Communications Manger Administration.

https://

2. Under Media Resources-> Music On Hold Audio Source, upload the .wav file and save it. This process generates the G.729 format.

3. Log in using the Secure Shell (SSH) Protocol with the Cisco Unified OS Administration username and password to access the command-line interface.

* To list the moh files:-
admin:file list activelog mohprep/*

1041.alaw.wav 1041.g729.wav
1041.ulaw.wav 1041.wb.wav
1041.xml 1061.alaw.wav
1061.g729.wav 1061.ulaw.wav
1061.wb.wav 1061.xml
CiscoMOHSourceReport.xml SampleAudioSource.alaw.wav
SampleAudioSource.g729.wav SampleAudioSource.ulaw.wav
SampleAudioSource.wb.wav SampleAudioSource.xml
dir count = 0, file count = 16

4. Execute the command file get activelog mohprep/.g729.wav and specify the Secure FTP (SFTP) server details to fetch this file.

N.B. I used freeFTPd for SFTP and putty for ssh terminal
and on the UCM ssh run the following command:-

admin:file get activelog mohprep/1061.g729.wav
Please wait while the system is gathering files info …done.
Sub-directories were not traversed.
Number of files affected: 1
Total size in Bytes: 3510
Total size in Kbytes: 3.4277344
Would you like to proceed [y/n]? y

SFTP server IP:10.1.1.1 (SFTP server ip on which freeFTPd running)
SFTP server port [22]:
User ID:mohuser (user created on freeFTPD – sftp)
Password: *****

Directory: /

.
Transfer completed.
admin:

SIP Trunk Diversion Header Modification

April 28, 2011

Certain Service Providers expect the call originated from theirs own validated DID ANI. And if we need to make a call originated from non validated number through the SIP trunk then we need to use a Diversion header to bypass the limitation.
Eg.

- Phone A’s DID is 4108887777 which is registered to Service provider A
– SIP service is with Service Provider B which allocates DID range of 202888XXXX
when phone A makes an outbound call then the call gets dropped at Service provider B sip server. This is because the SIP server is expecting the call originated from the specified range only 202888XXXX

voice class sip-profiles 2
request INVITE sip-header Diversion add “Diversion: <sip:2028881111@>;privacy=off;reason=deflection;screen=yes”

dial-peer voice 100 voip
destination-pattern [2-9]..[2-9]……
voice-class codec 1
voice-class sip g729 annexb-all
voice-class sip early-offer forced
voice-class sip profiles 2
session protocol sipv2
session target ipv4:
dtmf-relay rtp-nte sip-notify
no vad

a.a.a.a = local CUBE router IP
xx.xx.xx.xx = service provider SIP server
debug ccsip messages

INVITE sip:+1234567890@XX.XX.XX.XX:5060 SIP/2.0
Via: SIP/2.0/UDP a.a.a.a:5060;branch=z9hG4bK121DBAB28
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
From: ;tag=E4492450-FF1
To:
Date: Thu, 28 Apr 2011 19:03:31 GMT
Call-ID: AD48393-710111E0-9564EA07-56006A3D@10.7.1.4
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 181619371-1895895520-2506025479-1442867773
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Timestamp: 1304017411
Contact:
Call-Info: ;method=”NOTIFY;Event=telephone-event;Duration=2000″
Diversion: ;privacy=off;reason=deflection;screen=yes
Max-Forwards: 69
Session-Expires: 86400
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 258
Diversion: ;

Hello world!

April 28, 2011

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