Solved: FXS without dial tone in CME - Page 2

ivan aguilar · ‎08-20-2012

Hi!

I have a problem configuring a FXS port in a router 2911 with CME.

After introduce all commands for the configuration, i have not a dial tone, Now the router is not connected with any BRI or PRI, but I think that I should hear a dial tone when an analogic phone where connected, these are the commands:

Voice ports

voice-port 0/1/0
signal groundStart
no echo-cancel enable
echo-cancel mode 1
cptone ES
timeouts ringing infinity
connection plar 2000
bearer-cap Speech
caller-id enable
!
voice-port 0/1/1
signal groundStart
no echo-cancel enable
echo-cancel mode 1
cptone ES
timeouts ringing infinity
connection plar 2000
bearer-cap Speech
caller-id enable

SCCP

sccp local GigabitEthernet0/0.10
sccp ccm x.x.x.x identifier 1 version 7.0
sccp

sccp ccm group 1
associate ccm 1 priority 1

Dial-peer association

dial-peer voice 500 pots
service stcapp
port 0/1/0
!
dial-peer voice 501 pots
service stcapp
port 0/1/1

Do i need anything else??

Thanks!!!!

Ayodeji Okanlawon · ‎08-21-2012

I have used the DSP calcultaor to check how many DSPs you need for your current configuration. 30 E1 channels, 2 High codec complexity conference session and 2 FXS ports. The result came up as 4 DSPs..i.e pvdm3-64. From your sh diag, you only have 3 dsps, i.e pvdm3-16 + pvdm3-32.

What you can do to confirm if this is a PVDM related issue is to shutdown the conference dspfarm profile. That will free up dsps. You can test the analogue ports again. If it works this time, then you know you need more DSPs

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ivan aguilar · ‎08-21-2012

Hi!

I shutdown the conference dspfarm profile but stills fails, may be is other thing:

Router(config)#dspfarm profile 1
Router(config-dspfarm-profile)#shut

Disabling profile will disconnect active CONFERENCING calls,
do you want to continue ? [yes/no]yes

Still I receive a busy tone , please could you send me link the the DSP calculator? and, other question why pvdm3-16 + pvdm3-32 give me 3 DSP? and 3-64 give me 4?

Thanks!!

Ayodeji Okanlawon · ‎08-21-2012

Hi, Here is the explanation:

PVDM3-16 = 1dsp

PVDM3-32= 2 dsp

PVDM3-48= 3dsp

PVDM3-64=4dsp

http://www.cisco.com/web/applicat/dsprecal/dsp_calc.html

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ivan aguilar · ‎08-21-2012

Thanks for your help!

Could resolve this issue if I upgrade the IOS?

regards!

arunkum3 · ‎08-21-2012

Hi Ivan,

Can you try this config to test out:

Remove all the commands under the voice-port:

voice-port 0/1/0

!

voice-port 0/1/1

Dial-peer association should look like this (remove the extra config)

dial-peer voice 500 pots

destination-p 1001

port 0/1/0

!

dial-peer voice 501 pots

destination-p 1002

port 0/1/1

Save config and reload the router if required so that there are no cached values.

Basically any phone should have a line appearence (which is provided by the destination-pattern command on an FXS port. This config is missing in your "show run".

Let me know if this resolves the issue.

Regards,

Arun Kumar MV

paolo bevilacqua · ‎08-21-2012

Save config and reload the router if required so that there are no cached values.

Cisco router do not need reload after configuration changes.

Basically any phone should have a line appearence (which is provided by the destination-pattern command on an FXS port. This config is missing in your "show run".

That does not create a line appearancebut only support for basic calling. True line appearance for analog phones in CME is done configuring the port in SCCP mode, that is what the OP had , at least partially, in the initial posted config

rofallar · ‎08-21-2012

If the voice port turns up you have enough dsp resources. Your CME config is incomplete. First make sure the ip of the Sccp service points to the source address under telephony service. Next show the stcapp device names it's either under stcapp app or Sccp I don't remember off the top of my head. After you get the name add an ephone of type analog and whatever Mac address you got from the last step. Finally add an ephone-dn and assign it to that ephone.

Sent from Cisco Technical Support iPad App

ivan aguilar · ‎08-21-2012

Hi arunkum3 !!

Now I have a dial-tone, even with this configuration in voice-port:

voice-port 0/1/0

no echo-cancel enable

echo-cancel mode 1

cptone ES

timeouts ringing infinity

caller-id enable

!

voice-port 0/1/1

no echo-cancel enable

echo-cancel mode 1

cptone ES

timeouts ringing infinity

caller-id enable

Question is about destination-p in the dial-peer, i have tested when i quit the command ´service stcapp´ and put destinattion-pattern runs, so the problem was there. my question are:

Why do i must to quit this? in cisco.com explains this command is mandatory

The destinattion-pattern 1001 and 1002 why? is as example, can i put there whatever?

thanks and regards!!!!!

paolo bevilacqua · ‎08-21-2012

With service stcapp, you have the port in SCCP mode.

With destination-pattern, in regular dial-peer mode.

The two modes, and commands are mutually exclusive.

arunkum3 · ‎08-21-2012

Hi Ivan,

Paolo is right. We don't need the destination-pattern command if you are looking to control the analog endpoints via CME. The destination-pattern is for providing an extension number to the analog stations.

However you may use the destination-pattern command to verify the hardware (VIC3 card).

So the correct configuration would be:

voice-port 0/1/0

caller-id enable

cptone ES

voice-port 0/1/1

caller-id enable

cptone ES

stcapp ccm-group 1

stcapp

sccp local GigabitEthernet0/0.10

sccp ccm x.x.x.x identifier 1

sccp

sccp ccm group 1

associate ccm 1 priority 1

dial-peer voice 500 pots

 service stcapp

port 0/1/0

dial-peer voice 501 pots

 service stcapp

port 0/1/1
 
telephony-service
no auto-reg-ephone
max-ephones 50
max-dn 50
ip source-address x.x.x.x port 2000


 
Other example configuration:
 http://www.cisco.com/en/US/docs/ios/voice/fxs/configuration/guide/fxsbasic_ps10592_TSD_Products_Configuration_Guide_Chapter.html#wp1010513

I came across a Cisco Documentation that says that while connecting the RJ11 cable to the VIC3-2FXS/DID, the router has to be powered off:

http://www.cisco.com/en/US/partner/docs/routers/access/interfaces/ic/hardware/installation/guide/2port_FXS_DID_VIC.html#wp1065733

So first enter the configuration commands. Power off the router, connect the RJ11 cable to the phone. Power on the router and check if the Phone has the dial tone.

Regards,

Arun Kumar MV

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ivan aguilar · ‎08-22-2012

Hi again!

I did the changes you said me but when I put ´service stcapp´ in a dial-peer sounds always a busy tone in the analog phone, the finally configuration is as below:

RouterMarpetrol#sh run
Building configuration...

!
card type e1 0 0
no logging console
enable secret 4 Tw/TVweCrKP8tReUhD.qT6x6FACDOS52YTSZ8LoMV5A
!
no aaa new-model
clock timezone ES 1 0
clock summer-time VERANO date Jun 21 2012 0:00 Sep 20 2012 0:00
network-clock-participate wic 0
!
no ipv6 cef
ip source-route
ip cef
!
!
multilink bundle-name authenticated
!
!
stcapp ccm-group 1
stcapp
!
!
isdn switch-type primary-net5
!
crypto pki token default removal timeout 0
!
!
voice-card 0
dspfarm
dsp services dspfarm
!
!
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none
h323
call start slow
sip
redirect contact order best-match
!
voice class codec 100
codec preference 1 g711ulaw
codec preference 2 g711alaw
codec preference 3 g729r8
!

!
!
license udi pid CISCO2911/K9 sn FCZ162070T9
license accept end user agreement
license boot module c2900 technology-package uck9
hw-module pvdm 0/0
!
hw-module pvdm 0/1
!
!

redundancy
!
!
controller E1 0/0/0
pri-group timeslots 1-31
description PRIMARIO DE VOZ
!

!
interface Embedded-Service-Engine0/0
no ip address
shutdown
!
interface GigabitEthernet0/0
no ip address
duplex auto
speed auto
!
interface GigabitEthernet0/0.10
encapsulation dot1Q 10
ip address 10.7.5.4 255.255.255.0
!
interface GigabitEthernet0/0.20
encapsulation dot1Q 20
ip address 212.4.110.10 255.255.255.0
!
interface GigabitEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
interface GigabitEthernet0/2
ip address 192.168.10.1 255.255.255.0
duplex auto
speed auto
!
interface Serial0/0/0:15
no ip address
encapsulation hdlc
isdn switch-type primary-net5
isdn incoming-voice voice
no cdp enable
!
ip forward-protocol nd
!
ip http server
ip http authentication local
no ip http secure-server
ip http path flash:/gui
!

!
!
!
control-plane
!
!
voice-port 0/0/0:15
no echo-cancel enable
echo-cancel mode 1
cptone ES
connection plar 2000
shutdown
bearer-cap Speech
!
voice-port 0/1/0
no echo-cancel enable
echo-cancel mode 1
cptone ES
timeouts ringing infinity
caller-id enable
!
voice-port 0/1/1
no echo-cancel enable
echo-cancel mode 1
cptone ES
timeouts ringing infinity
caller-id enable
!
!
!
mgcp profile default
!
sccp local GigabitEthernet0/0.10
sccp ccm 10.7.5.4 identifier 1 version 7.0
sccp
!
sccp ccm group 1
associate ccm 1 priority 1
!
dspfarm profile 2 transcode
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
maximum sessions 2
associate application SCCP
shutdown
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2
associate application SCCP
shutdown
!
dspfarm profile 3 mtp
codec g711ulaw
maximum sessions hardware 3
shutdown
!
dial-peer voice 1 pots
translation-profile incoming PSTN
preference 1
incoming called-number .
port 0/0/0:15
!
dial-peer voice 2 pots
description salientes a moviles
translation-profile incoming PSTN
destination-pattern 6........
port 0/0/0:15
forward-digits all
!
dial-peer voice 100 pots
description llamadas internacionales
destination-pattern 0T
port 0/0/0:15
forward-digits all
!
dial-peer voice 500 pots
service stcapp
port 0/1/0
!
dial-peer voice 501 pots
service stcapp
port 0/1/1
!
!
!
!
gatekeeper
shutdown
!
!
telephony-service
no auto-reg-ephone
max-ephones 50
max-dn 50
ip source-address 10... port 2000
max-redirect 20
system message Marpetrol
user-locale ES
network-locale ES
load 7942 sccp
load 7962 sccp
time-format 24
date-format dd-mm-yy
keepalive 15
max-conferences 8 gain -6
moh music-on-hold.au
web admin customer name Petrol password Petrol
transfer-system full-consult
transfer-pattern ....
secondary-dialtone 0
xmltest
create cnf-files version-stamp 7960 Aug 22 2012 10:24:30

The webpage you send me is the same pdf file from cisco i used, so the first configuration was fine, the only runs is change stcapp in the dial-peer for destination-pattern.

As i will use this with a CME, is there any problem if I live the dial-peer with destination-p in order to get it runs instead of service stcapp??

And the destination-p number could be any number? or must be the telephony number of the analog phone?

Thanks Arunkum and Paolo!

ivan aguilar · ‎08-23-2012

Hi!

All runs properly now, the destinattion-p number is used as the extension of the analog phone.

So question is, what is used for the ´stcapp service´ ???

thanks!

Un saludo.

paolo bevilacqua · ‎08-24-2012

To have an analog phone access features that otherwise it doesn't have.

Thanks for the nice rating and good luck!