01-25-2008 03:28 AM - edited 03-15-2019 08:25 AM
Hi All,
My customer with using CCME complaint that the post-dial delay is too long for PSTN calls (they have pressed #). Actually we found that the time is more or less the same using CCM. I compared the delay with Notel and Avaya IP Phone system today and found that Cisco is really the slowest with obvious much longer time compared with Avaya. Nortel is the second, Avaya is the fastest. The test is using IP phones of different brands to dial to my mobile. The voice gateway is H323 gateway (except CCME combined with Voice Gateway). Outgoing line using FXO. Can I do something to shorten the delay?
Thank you!
Best Regards,
Teru Lei
01-25-2008 03:47 AM
Hi, if they press #, the router calls immediately. You can collect "debug vpm signal" with ms timestamps, to confirm.
Hope this helps, please rate post if it does!
01-25-2008 07:05 AM
I have tried by myself. If not press #, as we all know, the interdigit timeout will make the delay even longer.I think it's not CCM or CCME problem. It's the voice gateway.... I can see the signaling go through... but really slow to make the call setup. then I go to other customers' site who use Notel and Avaya system to test, the speed is very faster, specially for Avaya, very fast connection. But the Notel and Avaya systems are not implemented by my company so that I just can make call to test. And what I know is that the customers all use FXO for outgoing calls.
01-25-2008 07:09 AM
Ok, are you using MGCP or H.323 ?
If the latter, you can collect "debug vpm signal" after configuring "service timestamps debug datetime msec localtime", one can check where the delay accumulates and if any timer can be shortened to reduce it.
01-26-2008 12:32 AM
Thanks for your reply. I am using H323. actually I have several clients using CCM/CCME with Cisco Voice Gateway but no complaint but now one customer complaint then I tried to compare different brand IP phone system and unhappy can this result... is there any common timmers I can use to optimized the delay?
Thank you!
Best Regards,
Teru Lei
01-26-2008 03:05 AM
Possibly "timeouts initial" under voice-port.
Try to be present when your customer do tests, because sometime turns out that beside inadequate testing procedures are used, critiques on minor issues like that, are to blame cisco for whatever reason, as they have already preference for some other vendor.
And in reality, no organization that really cares about call quality and features uses analog lines, but ISDN exclusively.
01-26-2008 11:29 PM
Thanks! I will try once I get chance to do that
01-27-2008 06:55 PM
Hi All,
Here's some output from debug vpm signal:
*Jan 28 02:02:01: htsp_process_event: [50/0/52.1, EFXS_ONHOOK, E_DSP_SIG_1100]efxs_onhook_offhook htsp_setup_ind
*Jan 28 02:02:01: [50/0/52.1] get_local_station_id calling num=85983332 calling name=Steven Chan calling time=01/28 10:02 orig called=
*Jan 28 02:02:01: htsp_process_event: [50/0/52.1, EFXS_WAIT_SETUP_ACK, E_HTSP_SETUP_ACK]efxs_check_auto_call
*Jan 28 02:02:02: htsp_digit_ready(50/0/52.1): digit = 9
*Jan 28 02:02:02: htsp_call_bridged invoked
*Jan 28 02:02:02: htsp_process_event: [1/1/3, FXOLS_CONNECT, E_HTSP_VOICE_CUT_THROUGH]fxols_connect_proc_voice
*Jan 28 02:02:03: htsp_digit_ready(50/0/52.1): digit = 3
*Jan 28 02:02:03: htsp_digit_ready(50/0/52.1): digit = 9
*Jan 28 02:02:03: htsp_digit_ready(50/0/52.1): digit = 8
*Jan 28 02:02:04: htsp_digit_ready(50/0/52.1): digit = 9
*Jan 28 02:02:04: htsp_digit_ready(50/0/52.1): digit = 1
*Jan 28 02:02:04: htsp_digit_ready(50/0/52.1): digit = 5
*Jan 28 02:02:04: htsp_digit_ready(50/0/52.1): digit = 3
*Jan 28 02:02:06: htsp_digit_ready(50/0/52.1): digit = #
*Jan 28 02:02:06: htsp_timer_stop3
*Jan 28 02:02:06: htsp_process_event: [50/0/52.1, EFXS_OFFHOOK, E_HTSP_PROCEEDING]efxs_offhook_proceeding
*Jan 28 02:02:06: [50/0/52.1] set signal state = 0x8 timestamp = 0htsp_setup_req
*Jan 28 02:02:06: htsp_timer - 1300 msec
htsp_call_feature: caller id enable 0x3 call_connected 1
*Jan 28 02:02:10: htsp_process_event: [50/0/52.1, EFXS_OFFHOOK, E_HTSP_CONNECT]efxs_offhook_connect
*Jan 28 02:02:10: [50/0/52.1] set signal state = 0x6 timestamp = 0
*Jan 28 02:02:10: htsp_process_event: [50/0/52.1, EFXS_CONNECT, E_HTSP_CALLERID_WAITING]
*Jan 28 02:02:10: efxs_callerid_update
*Jan 28 02:02:10: efxs_callerid_update process caller_id_string
*Jan 28 02:02:10: efxs_callerid_update process caller_id_string OK
*Jan 28 02:02:10: efxs_callerid_update number= [3989153] name= []
Seems that it gets around 10 sec to setup the call, and also the voice port send digit to outside is slow.(02:02:01 to 02:02:10) any timers I should try to tune the setup time?
Thank you very much!
Best Regards,
Teru Lei
01-27-2008 09:22 PM
Post your router configuration - minus IP Addresses and passwords.
01-27-2008 11:46 PM
Thanks! Part 1:
!
version 12.4
service timestamps debug datetime
service timestamps log datetime
no service password-encryption
!
hostname xxxxx
!
boot-start-marker
boot system flash:c3845-spservicesk9-mz.124-9.T7.bin
boot-end-marker
!
logging buffered 51200 warnings
!
no aaa new-model
!
resource policy
!
clock timezone CHI 8
no network-clock-participate slot 1
no network-clock-participate slot 2
no ip dhcp use vrf connected
ip dhcp excluded-address xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx
!
ip dhcp pool ITS
network xxx.xxx.xxx.xxx 255.255.255.0
option 150 ip xxx.xxx.xxx.xxx
default-router xxx.xxx.xxx.xxx
!
!
!
ip cef
!
!
ip domain name xxxx.com
ip multicast-routing
!
voice-card 0
no dspfarm
!
voice-card 1
no dspfarm
!
voice-card 2
no dspfarm
!
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
no h225 timeout keepalive
!
!
!
!
!
!
!
!
!
!
!
!
!
voice translation-rule 1
rule 1 /^859836/ /36/
rule 2 /^859833/ /33/
rule 3 /^63/ /33/
!
voice translation-rule 2
rule 1 /^87/ //
rule 2 /^82/ //
!
voice translation-rule 3
rule 1 /^91/ /1/
rule 2 /^92/ /2/
rule 3 /^93/ /3/
rule 4 /^94/ /4/
rule 5 /^95/ /5/
rule 6 /^96/ /6/
rule 7 /^97/ /7/
rule 8 /^98/ /8/
rule 9 /^99/ /9/
!
!
voice translation-profile Intersite
translate called 2
!
voice translation-profile VM
translate calling 1
translate called 1
translate redirect-called 1
!
voice translation-profile external-local
translate called 3
!
!
!
!
!
!
01-27-2008 11:47 PM
Part 2:
interface GigabitEthernet0/0
description $ETH-LAN$$ETH-SW-LAUNCH$$INTF-INFO-GE 0/0$
ip address xxx.xxx.xxx.xxx 255.255.255.0
duplex auto
speed auto
media-type rj45
!
interface Service-Engine4/0
ip unnumbered GigabitEthernet0/0
service-module ip address xxx.xxx.xxx.xxx 255.255.255.0
service-module ip default-gateway xxx.xxx.xxx.xxx
!
ip route 0.0.0.0 0.0.0.0 xxx.xxx.xxx.xxx
ip route xxx.xxx.xxx.xxx 255.255.255.255 Service-Engine4/0
!
!
ip http server
ip http authentication local
ip http secure-server
ip http timeout-policy idle 60 life 86400 requests 10000
ip http path flash:
!
!
!
tftp-server flash:P00303020214.bin
tftp-server flash:P00305000301.sbn
tftp-server flash:P00403020214.bin
tftp-server flash:P0030702T023.bin
tftp-server flash:P0030702T023.loads
tftp-server flash:P0030702T023.sb2
tftp-server flash:P0030702T023.sbn
tftp-server flash:P00405000700.bin
tftp-server flash:P00405000700.sbn
!
control-plane
!
!
!
voice-port 0/1/0
operation 4-wire
type 5
signal immediate
!
voice-port 0/1/1
operation 4-wire
type 5
signal immediate
!
voice-port 0/2/0
operation 4-wire
type 5
signal immediate
!
voice-port 0/2/1
operation 4-wire
type 5
signal immediate
!
voice-port 0/3/0
operation 4-wire
type 5
signal immediate
shutdown
!
voice-port 0/3/1
operation 4-wire
type 5
signal immediate
shutdown
!
voice-port 1/0/0
!
voice-port 1/0/1
connection plar 3690
!
voice-port 1/0/2
connection plar 3690
!
voice-port 1/0/3
connection plar 3690
!
voice-port 1/1/0
connection plar 3308
!
voice-port 1/1/1
connection plar 6002
!
voice-port 1/1/2
connection plar 3690
!
voice-port 1/1/3
connection plar 3690
!
voice-port 2/0/0
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/0/1
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/0/2
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/0/3
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/1/0
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/1/1
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/1/2
signal did immediate
input gain 7
description Incoming Trunk from External
!
voice-port 2/1/3
signal did immediate
input gain 7
description Incoming Trunk from External
!
!
!
!
dial-peer cor custom
name 82
name 87
name 00
name none
name 9[1-9]
!
!
dial-peer cor list css-00
member 00
!
dial-peer cor list css-82
member 82
!
dial-peer cor list css-87
member 87
!
dial-peer cor list css-00-82
member 82
member 00
!
dial-peer cor list css-00-87
member 87
member 00
!
dial-peer cor list css-82-87
member 82
member 87
01-27-2008 11:48 PM
Part 3:
!
dial-peer cor list css-00-82-87
member 82
member 87
member 00
!
dial-peer cor list css-none
member none
!
!
dial-peer voice 99 voip
translation-profile outgoing VM
destination-pattern 369[014]
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 98 voip
translation-profile outgoing VM
destination-pattern 8598369[014]
session protocol sipv2
session target ipv4:xxx.xxx.xxx.xxx
dtmf-relay sip-notify
codec g711ulaw
no vad
!
dial-peer voice 2 pots
corlist outgoing css-82
translation-profile outgoing Intersite
destination-pattern 82.T
no digit-strip
port 0/1/0
!
dial-peer voice 3 pots
corlist outgoing css-82
translation-profile outgoing Intersite
destination-pattern 82.T
no digit-strip
port 0/1/1
!
dial-peer voice 4 pots
corlist outgoing css-82
translation-profile outgoing Intersite
destination-pattern 82.T
no digit-strip
port 0/2/0
!
dial-peer voice 5 pots
corlist outgoing css-82
translation-profile outgoing Intersite
destination-pattern 82.T
no digit-strip
port 0/2/1
!
dial-peer voice 8 pots
translation-profile outgoing external-local
destination-pattern 9[1-9]T
port 1/0/0
forward-digits 8
!
dial-peer voice 9 pots
translation-profile outgoing external-local
preference 2
destination-pattern 9[1-9]T
port 1/0/1
forward-digits 8
!
dial-peer voice 10 pots
translation-profile outgoing external-local
preference 2
destination-pattern 9[1-9]T
port 1/0/2
forward-digits 8
!
dial-peer voice 11 pots
translation-profile outgoing external-local
preference 2
destination-pattern 9[1-9]T
port 1/0/3
forward-digits 8
!
dial-peer voice 13 pots
translation-profile outgoing external-local
preference 3
destination-pattern 9[1-9]T
port 1/1/1
forward-digits 8
!
dial-peer voice 14 pots
translation-profile outgoing external-local
destination-pattern 9[1-9]T
port 1/1/2
forward-digits 8
!
dial-peer voice 15 pots
translation-profile outgoing external-local
destination-pattern 9[1-9]T
port 1/1/3
forward-digits 8
!
dial-peer voice 22 pots
corlist outgoing css-87
translation-profile outgoing Intersite
destination-pattern 87.T
no digit-strip
port 0/1/0
!
dial-peer voice 23 pots
corlist outgoing css-87
translation-profile outgoing Intersite
destination-pattern 87.T
no digit-strip
port 0/1/1
!
dial-peer voice 24 pots
corlist outgoing css-87
translation-profile outgoing Intersite
destination-pattern 87.T
no digit-strip
port 0/2/0
!
01-27-2008 11:48 PM
Part 4:
dial-peer voice 25 pots
corlist outgoing css-87
translation-profile outgoing Intersite
destination-pattern 87.T
no digit-strip
port 0/2/1
!
dial-peer voice 38 pots
corlist outgoing css-00
description Route to IDD
destination-pattern 900T
port 1/0/0
prefix 00
!
dial-peer voice 39 pots
corlist outgoing css-00
description Route to IDD
preference 2
destination-pattern 900T
port 1/0/1
prefix 00
!
dial-peer voice 40 pots
corlist outgoing css-00
description Route to IDD
preference 2
destination-pattern 900T
port 1/0/2
prefix 00
!
dial-peer voice 41 pots
corlist outgoing css-00
description Route to IDD
preference 2
destination-pattern 900T
port 1/0/3
prefix 00
!
dial-peer voice 43 pots
corlist outgoing css-00
description Route to IDD
preference 3
destination-pattern 900T
port 1/1/1
prefix 00
!
dial-peer voice 44 pots
corlist outgoing css-00
description Route to IDD
destination-pattern 900T
port 1/1/2
prefix 00
!
dial-peer voice 45 pots
corlist outgoing css-00
description Route to IDD
destination-pattern 900T
port 1/1/3
prefix 00
!
dial-peer voice 49 pots
corlist outgoing css-00
description Route to IDD Alias
preference 2
destination-pattern 901T
port 1/0/1
prefix 01
!
dial-peer voice 50 pots
corlist outgoing css-00
description Route to IDD Alias
preference 2
destination-pattern 901T
port 1/0/2
prefix 01
!
dial-peer voice 51 pots
corlist outgoing css-00
description Route to IDD Alias
preference 2
destination-pattern 901T
port 1/0/3
prefix 01
!
dial-peer voice 53 pots
corlist outgoing css-00
description Route to IDD Alias
preference 3
destination-pattern 901T
port 1/1/1
prefix 01
!
dial-peer voice 54 pots
corlist outgoing css-00
description Route to IDD Alias
destination-pattern 901T
port 1/1/2
prefix 01
!
dial-peer voice 55 pots
corlist outgoing css-00
description Route to IDD Alias
destination-pattern 901T
port 1/1/3
prefix 01
!
dial-peer voice 48 pots
corlist outgoing css-00
description Route to IDD Alias
destination-pattern 901T
port 1/0/0
prefix 01
!
dial-peer voice 1000 pots
description For Boss Phone
destination-pattern 5T
port 1/1/0
!
!
!
!
telephony-service
load 7910 P00405000700
load 7960-7940 P0030702T023
max-ephones 200
max-dn 400
ip source-address xxx.xxx.xxx.xxx port 2000
max-redirect 7
auto assign 1 to 100
no service directed-pickup
system message Your current options
url services http://xxx.xxx.xxx.xxx/voiceview/common/login.do
url authentication http://xxx.xxx.xxx.xxx/voiceview/authentication/authenticate.do
time-zone 42
time-format 24
dialplan-pattern 1 xxxxx... extension-length 4
voicemail 3691
max-conferences 3 gain -6
call-forward pattern .T
moh flash:music-on-hold.au
multicast moh 239.1.1.1 port 16384
web admin system name admin password password
dn-webedit
time-webedit
transfer-system full-consult
secondary-dialtone 9
login timeout 120
after-hours block pattern 1 90
after-hours block pattern 2 9+
after-hours block pattern 10 5
after-hours day Sun 00:00 23:59
after-hours day Mon 00:00 23:59
after-hours day Tue 00:00 23:59
after-hours day Wed 00:00 23:59
after-hours day Thu 00:00 23:59
after-hours day Fri 00:00 23:59
after-hours day Sat 00:00 23:59
create cnf-files version-stamp Jan 01 2002 00:00:00
01-27-2008 11:49 PM
Thank you! and I have removed ephone-dn and ephone config
01-28-2008 12:02 AM
Two solutions -
1) Make your destination patterns specific to the PSTN number ranges. For example, if your local city calls are 8 digits, in the range of 5XXX XXXX - 8XXX XXXX, then create a dial peer like this -
!
dial-peer voice 111 pots
description - local calls
destination-pattern 9[5-8].......
port 0/0/0
!
you can also use trunk groups to reduce the quantity dial peers by group the voice ports into trunk groups and using these on the dial peers. Do a search on netpro for config examples.
You will to create multiple dial peers to cover your full internal/local/regional numbering plan. Use a destination pattern of 9011T for international calls.
2) Keep your current dial peer config and reduce the interdigit timeout. The default is 10 seconds -
telephony-service
timeouts interdigit 3
Don't reduce it any more than 3 seconds as you find that people tend to hesitate when dialing and the timeout is too short the dialing starts before they have finished scratching their noses !
Find answers to your questions by entering keywords or phrases in the Search bar above. New here? Use these resources to familiarize yourself with the community: