Cisco CCM, CCME, PSTN post-dial delay longer than other brands

Unanswered Question
Jan 25th, 2008

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

I have this problem too.
0 votes
  • 1
  • 2
  • 3
  • 4
  • 5
Overall Rating: 0 (0 ratings)
Loading.
Paolo Bevilacqua Fri, 01/25/2008 - 03:47

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!

teru-lei Fri, 01/25/2008 - 07:05

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.

Paolo Bevilacqua Fri, 01/25/2008 - 07:09

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.

teru-lei Sat, 01/26/2008 - 00:32

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

Paolo Bevilacqua Sat, 01/26/2008 - 03:05

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.

teru-lei Sun, 01/27/2008 - 18:55

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

teru-lei Sun, 01/27/2008 - 23:46

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

!

!

!

!

!

!

teru-lei Sun, 01/27/2008 - 23:47

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

teru-lei Sun, 01/27/2008 - 23:48

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

!

teru-lei Sun, 01/27/2008 - 23:48

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

teru-lei Sun, 01/27/2008 - 23:49

Thank you! and I have removed ephone-dn and ephone config

pcameron Mon, 01/28/2008 - 00:02

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 !

Paolo Bevilacqua Mon, 01/28/2008 - 02:37

Hi,

digit collection ended at 02:02:06. Then I don't see anything related to the FXO port ?!?

teru-lei Mon, 01/28/2008 - 17:10

No error or adnormal in the message in side. Just seems the delay is too long

Paolo Bevilacqua Tue, 01/29/2008 - 02:12

Cannot tell because there is not engough infor about FXO activity in your trace.

Do the measurement yourself, go off hook with 9 and get dialtone. dial, and measure from last digit to first ringback. No # is necessary in this case.

Repeat, but this time compose the number with the phone on-hook, followed by #. The press dial softkey and measure to first ringback.

The difference should be at most, 1 or 2 seconds.

Let us know.

Actions

This Discussion