Wednesday, November 10, 2010

Cisco Phone Designer

When trying to preview wallpapers or ring tones on a 7941 phone your receive "An unknown error occurred on your Cisco IP Phone". Authentication with a user / password appears to work fine at loggin because the available phone list is updated appropriately.

It appears in my case the CUCM Authentication URL was incorrect. Typically it's http://YourCUCMserver:8080/ccmcip/authenticate.jsp by default (7.1.X). Whoever was testing the IP paging solution here never set it back to default and all push technologies likely failed.

Check your Enterprise Setting URLs, specifically the Authentication URL. Remember to to reboot phones after modifying this entry to have it apply.

Sunday, November 07, 2010

SRST sample - MGCP PRI with DIDs

Here is a sample of a MGCP controlled gateway with a working SRST configuration. The interesting element is the translation rule to perform DID manipulation in fall back mode:

version 12.4
service timestamps debug datetime msec
service timestamps log datetime localtime show-timezone
no service password-encryption
boot system flash c2801-spservicesk9-mz.124-24.T3.bin
card type t1 0 0
card type t1 0 2
logging message-counter syslog
logging buffered 32768
enable secret 5 -removed-
no aaa new-model
clock timezone EDT -4
network-clock-participate wic 0
network-clock-participate wic 2
network-clock-select 2 T1 0/2/0
network-clock-select 3 T1 0/0/0
dot11 syslog
ip source-route
no ip dhcp use vrf connected
ip dhcp excluded-address
ip dhcp pool Site1Voice
option 150 ip
ip cef
ip name-server
ip name-server
ip multicast-routing
no ipv6 cef
multilink bundle-name authenticated
isdn switch-type primary-ni
trunk group FXO
voice service voip
fax protocol cisco
call preserve
voice class codec 1
codec preference 1 g711ulaw
voice class h323 1
h225 timeout tcp establish 1
voice translation-rule 1
rule 1 /^3910$/ /4101/
rule 2 /^3917$/ /4103/
rule 3 /^3918$/ /4104/
rule 4 /^3926$/ /4102/
rule 5 /^2800$/ /4100/
rule 6 /^39\(..\)$/ /41\1/
voice translation-profile ToSite1ForSRST
translate called 1
voice-card 0
dsp services dspfarm
service alternate default
username -removed- privilege 15 password 0 -removed-
log config
controller T1 0/0/0
cablelength long 0db
ds0-group 2 timeslots 13-24 type e&m-delay-dial
description -- LD T1 --
controller T1 0/2/0
cablelength long 0db
pri-group timeslots 1-24 service mgcp
description -- LOCAL PRI --
ip tftp source-interface FastEthernet0/0
class-map match-any VoIP-RTP-Trust
match ip dscp ef
class-map match-any VoIP-Control-Trust
match ip dscp cs3
match ip dscp af31
policy-map VOIP-Policy-Trust
class VoIP-RTP-Trust
priority percent 50
class VoIP-Control-Trust
bandwidth percent 10
class class-default
interface Loopback1
ip address
ip pim sparse-dense-mode
interface FastEthernet0/0
description -- to port 1 switch 2960 --
ip address
ip pim sparse-dense-mode
duplex auto
speed auto
no mop enabled
h323-gateway voip interface
h323-gateway voip bind srcaddr
interface FastEthernet0/1
description -- to port 24 switch 2960 --
ip address
duplex auto
speed auto
no mop enabled
interface Serial0/2/0:23
no ip address
encapsulation hdlc
isdn switch-type primary-ni
isdn incoming-voice voice
isdn bind-l3 ccm-manager
no cdp enable
interface Serial0/3/0
description MPLS Qwest
bandwidth 1536
ip address X.X.X.X
ip pim sparse-dense-mode
encapsulation ppp
service-policy output VOIP-Policy-Trust
router bgp 65XXX
no synchronization
bgp log-neighbor-changes
network mask
network X.X.X.X mask
neighbor X.X.X.X remote-as XXX
neighbor X.X.X.X soft-reconfiguration inbound
no auto-summary
ip forward-protocol nd
ip route
ip http server
no ip http secure-server
voice-port 0/0/0:2
echo-cancel coverage 64
voice-port 0/2/0:23
echo-cancel coverage 64
voice-port 0/1/0
description -Paging-
voice-port 0/1/1
ccm-manager fallback-mgcp
ccm-manager redundant-host
ccm-manager mgcp
no ccm-manager fax protocol cisco
ccm-manager music-on-hold
ccm-manager config server
ccm-manager config
mgcp call-agent 2427 service-type mgcp version 0.1
mgcp rtp unreachable timeout 1000 action notify
mgcp modem passthrough voip mode nse
mgcp package-capability rtp-package
mgcp package-capability sst-package
mgcp package-capability pre-package
no mgcp package-capability res-package
no mgcp timer receive-rtcp
mgcp sdp simple
mgcp fax t38 inhibit
mgcp bind control source-interface FastEthernet0/0
mgcp bind media source-interface FastEthernet0/0
mgcp profile default
sccp local FastEthernet0/0
sccp ccm identifier 1 version 7.0
sccp ccm group 999
bind interface FastEthernet0/0
associate ccm 1 priority 1
associate profile 1 register CONF_Site1
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 1
associate application SCCP
dial-peer voice 20 pots
destination-pattern 9[2-9]......
port 0/2/0:23
forward-digits 7
dial-peer voice 30 pots
destination-pattern 4125
port 0/1/0
forward-digits 0
dial-peer voice 999010 pots
service mgcpapp
port 0/1/0
dial-peer voice 999000991 pots
service mgcpapp
dial-peer voice 999000992 pots
service mgcpapp
port 0/0/0:2
dial-peer voice 1 pots
translation-profile incoming ToSite1ForSRST
incoming called-number .
port 0/2/0:23
dial-peer voice 21 pots
destination-pattern 91[2-9]..[2-9]......
port 0/2/0:23
forward-digits 11
dial-peer voice 911 pots
destination-pattern 911
port 0/2/0:23
forward-digits 3
dial-peer voice 9911 pots
destination-pattern 9911
port 0/2/0:23
forward-digits 3
secondary-dialtone 8
max-conferences 8 gain -6
transfer-system full-consult
timeouts interdigit 5
timeouts busy 30
timeouts ringing 60
ip source-address port 2000
max-ephones 25
max-dn 30 dual-line
system message primary Currently Running in Fall Back
system message secondary FallBack
default-destination 4100
moh Site1.AU
multicast moh port 16384 route
line con 0
line aux 0
line vty 0 4
password -removed-
line vty 5 15
password -removed-
scheduler allocate 20000 1000

Thursday, November 04, 2010

MGCP FXS port cannot break dial tone

Found user with butt set on an MGCP controlled VIC3-FXS/DID port can break DT and dial a number without issue. When a Sonitrol alarm panel is connected to the same port and dials the same number, either DT cannot be broken or number cannot be completed as dialed.

Fix was to reduce the input gain on the FXS port via CLI. This cannot be adjusted via CUCM like an FXO port, but is still an available option. Show voice port X/X/X indicates the default gain is 0db.

Working configuration is:

voice-port 0/2/1
input gain -6
description << Sonitrol Alarm >>
caller-id enable