01-29-2010 03:58 AM - edited 03-15-2019 09:14 PM
hi all,
not sure you can help me on this,
i have issue with Voice GW routing calls, issue like
1.normal PSTN call new office, i receive ring tone in the pstn phone and 7942 phone. however when i pick up the 7942 phone the PSTN phone just keep ringing almost like no one pickup the 7942 phone.
2. 7942 call out pstn, no ring tone in the 7942...my mobile phone get the office callerID and when i can pickup the phone but no voice on neither phones, then around 10 secs later, mymobile auto drop out. 7942 start with busy or bi bi bi bi signal.
here is some background.
i am installing a new office for our company which involve new gateway for the office.
i have setup the call manager part, which i have successfuly tested with Route list pointing to a existing gateway. everything fine.
any ideas?
thanks in advance.
here is a copy that very similar what it use in the new office.
vg#sh run
Building configuration...
Current configuration : 6345 bytes
!
version 12.4
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname
!
boot-start-marker
boot-end-marker
!
card type e1 0 0
logging buffered 51200 warnings
no logging console
enable secret 5
!
no aaa new-model
!
resource policy
!
network-clock-participate wic 0
network-clock-select 1 E1 0/0/0
!
!
ip cef
!
!
no ip domain lookup
ip domain name
ip multicast-routing
!
isdn switch-type primary-net5
voice-card 0
no dspfarm
dsp services dspfarm
!
!
!
no voice hunt unassigned-number
voice hunt user-busy
no voice hunt invalid-number
voice call disc-pi-off
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
fax protocol pass-through g711alaw
h323
!
!
voice class codec 10
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
codec preference 4 g729br8
!
!
!
voice class h323 10
h225 timeout tcp establish 3
no call preserve
!
!
!
!
!
!
!
!
!
!
voice translation-rule 2
!
!
voice translation-profile in-called
translate calling 3
translate called 2
!
voice translation-profile out-calling
translate calling 4
!
!
!
!
!
!
controller E1 0/0/0
pri-group timeslots 1-31
description PRI Circuit to TTT
!
!
!
!
interface GigabitEthernet0/0
description to XXXX
ip address 10.33.64.20 255.255.192.0
ip pim sparse-dense-mode
ip igmp join-group 239.1.1.1
duplex auto
speed auto
media-type rj45
h323-gateway voip interface
h323-gateway voip bind srcaddr 10.33.64.20
!
interface GigabitEthernet0/1
no ip address
duplex full
speed 100
media-type rj45
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
no logging event link-status
isdn switch-type primary-net5
isdn incoming-voice voice
isdn bchan-number-order ascending
isdn sending-complete
no cdp enable
!
ip route 0.0.0.0 0.0.0.0 10.33.64.1
!
!
no ip http server
ip http access-class 23
ip http authentication local
no ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip tacacs source-interface GigabitEthernet0/0
!
!
!
control-plane
!
!
!
voice-port 0/0/0:15
translation-profile incoming in-called
translation-profile outgoing out-calling
!
voice-port 0/1/0
no comfort-noise
cptone AU
timeouts interdigit 4
description FAX
!
voice-port 0/1/1
no comfort-noise
cptone AU
timeouts interdigit 4
description FAX
!
voice-port 0/1/2
no comfort-noise
cptone AU
timeouts interdigit 4
description FAX
!
voice-port 0/1/3
no comfort-noise
cptone AU
timeouts interdigit 4
description FAX
!
ccm-manager music-on-hold
!
!
sccp local GigabitEthernet0/0
sccp ccm 10.32.192.1 identifier 2
sccp ccm 10.32.192.2 identifier 1
sccp
!
sccp ccm group 1
associate ccm 2 priority 1
associate ccm 1 priority 2
associate profile 1 register CONF-VG-CRRL01
registration retries 7
registration timeout 8
keepalive retries 5
keepalive timeout 50
connect retries 5
switchback method immediate
switchback interval 15
!
dspfarm profile 1 conference
description IOS ConferenceBridge CONF-VG-CRRL01
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 4
associate application SCCP
!
!
dial-peer voice 200 voip
description To Subscriber
destination-pattern 41...
voice-class codec 10
voice-class h323 10
session target ipv4:10.32.192.2
dtmf-relay h245-alphanumeric
fax rate disable
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 201 voip
description to Publisher
preference 1
destination-pattern 41...
voice-class codec 10
voice-class h323 10
session target ipv4:10.32.192.1
dtmf-relay h245-alphanumeric
fax rate disable
ip qos dscp cs5 media
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 10 pots
preference 2
destination-pattern 0.T
progress_ind alert enable 8
incoming called-number .
direct-inward-dial
port 0/0/0:15
!
dial-peer voice 41023 pots
destination-pattern 41023
fax rate disable
port 0/1/0
!
dial-peer voice 41126 pots
destination-pattern 41126
fax rate disable
port 0/1/1
!
dial-peer voice 41128 pots
destination-pattern 41128
fax rate disable
port 0/1/2
!
dial-peer voice 41187 pots
destination-pattern 41187
fax rate disable
port 0/1/3
!
dial-peer voice 20 voip
voice-class codec 10
voice-class h323 10
incoming called-number .
no vad
!
!
!
!
call-manager-fallback
secondary-dialtone 0
max-conferences 4 gain -6
transfer-system full-consult
timeouts interdigit 5
ip source-address 10.33.64.20 port 2000
max-ephones 300
max-dn 600 dual-line
!
!
!
translation-profile outgoing out-calling
call-forward pattern .T
call-forward pattern 0.T
moh flash:moon.wav
multicast moh 239.1.1.1 port 16384 route 10.33.64.20 1.1.1.1
time-format 24
date-format dd-mm-yy
!
banner login ^CCCCCC
**********************************
Unauthorised access prohibited
any unauthorised access punishable
to maximum extent of the law
all activites on this device
are logged
**********************************
^C
!
line con 0
line aux 0
line vty 0 4
exec-timeout 30 0
login
!
scheduler allocate 20000 1000
ntp clock-period 17179936
ntp server 10.32.11.1 prefer
!
end
bris-vg#
02-01-2010 05:31 AM
The ring tone you hear when you pick up the phone is a recorded sound built into the firmware of the IP phone. Its NOT the PSTN tone.
You may have an overlap somewhere, which is why you do not hear the dial tone from PSTN.
Dial slowly and listen. For example:
Dial 9
What do you hear? (or outside number)
then the next number
next number
eventually it show up.
02-02-2010 03:41 PM
hi tcalinins
thanks for the reply,
i have tried dial slow and when i dial the outside number (0) here, i get
the 2nd dial tone. then keep dial the digits with no play back. once dial
the last digit, the pstn phone start to ring. (still no ringtone sound in
VOIP phone) however once i pick it up. both side have no sound.
i'll check the route plan again also GW see if anything obvious..
cheers
On Mon, Feb 1, 2010 at 11:31 PM, tcatlinins <
02-02-2010 08:17 PM
Sounds like you have a couple things...
When you dial the last digit, the PSTN gets the call, but still do not hear ring back. Since its h323, there seems to be a ring back issue on the h323 side.
Check this out
https://www.cisco.com/en/US/docs/ios/12_3/vvf_c/voice_troubleshooting/old/vts_h323.html
couple things to run through.
But also, since you are not hearing audio on either end, but the call is up, you have a codec transcoding issue. When the call is connected, hit the ? twice on the IP Phone. You shoudl be able to see the sending and receiving codec is going on.
02-02-2010 10:57 PM
thanks for all the help,
i think i miss to mention the GW was behind a VPN link.
and during another troubleshooting , some of the h323, skinny inspection has
been turned off. reenable the inspection all good now..
thanks
Jon
On Wed, Feb 3, 2010 at 2:17 PM, tcatlinins <
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