Analog phone with conference bridge software?

[Marcelo Bergamini]

Hi all!!

Could someone help me?

My customer has a VG204 and four analog phones. The phones are registered in CCM at a remote location. The phones are resgistered in a Device Pool with Media resource group e list correct.

The VG204 doesn't have DSPs.

Its possible the analogs phones to use CCM resouces, to complete conferences? There are some configuration required beyond the basic configuration?

[Marcelo Bergamini]

Hi Sadaseeven!

the phoen model is Conference Polycom.

The CCM version is 8.0.3.20000-2.

The output are below:

#show version
Cisco IOS Software, VG20X Software (VG20X-IPVOICE-M), Version 12.4(22)T5, RELEASE SOFTWARE (fc3)
Technical Support: http://www.cisco.com/techsupport
Copyright (c) 1986-2010 by Cisco Systems, Inc.
Compiled Wed 28-Apr-10 15:45 by prod_rel_team

ROM: System Bootstrap, Version 12.4(20r)YA1, RELEASE SOFTWARE (fc1)

TBR-VG20401-RXX_ITABORAI uptime is 4 weeks, 17 hours, 59 minutes
System returned to ROM by power-on
System image file is "flash:vg20x-ipvoice-mz.124-22.T5.bin"

Cisco VG204 (MPC8300) processor (revision 0x100) with 118784K/12288K bytes of memory.
Processor board ID FCH1426R007
MPC8300 CPU Rev: Part Number 0x8062, Revision ID 0x11
2 FastEthernet interfaces
4 Voice FXS interfaces
256K bytes of non-volatile configuration memory.
126000K bytes of ATA CompactFlash (Read/Write)

Configuration register is 0x2102

#####################################################################################

#show voice port 0/1

Foreign Exchange Station 0/1 Slot is 0, Port is 1
Type of VoicePort is FXS  ON-BOARD-4FXS
Operation State is DORMANT
Administrative State is UP
No Interface Down Failure
Description is FAX 1421
Noise Regeneration is enabled
Non Linear Processing is enabled
Non Linear Mute is disabled
Non Linear Threshold is -21 dB
Music On Hold Threshold is Set to -38 dBm
In Gain is Set to 0 dB
Out Attenuation is Set to 3 dB
Echo Cancellation is enabled
Echo Cancellation NLP mute is disabled
Echo Cancellation NLP threshold is -21 dB
Echo Cancel Coverage is set to 64 ms
Echo Cancel worst case ERL is set to 6 dB
Playout-delay Mode is set to adaptive
Playout-delay Nominal is set to 60 ms
Playout-delay Maximum is set to 1000 ms
Playout-delay Minimum mode is set to default, value 40 ms
Playout-delay Fax is set to 300 ms
Connection Mode is normal
Connection Number is not set
Initial Time Out is set to 15 s
Interdigit Time Out is set to 3 s
Call Disconnect Time Out is set to 0 s
Supervisory Disconnect Time Out is set to 750 ms
Ringing Time Out is set to infinity
Wait Release Time Out is set to 30 s
Companding Type is A-law
Region Tone is set for BR

Analog Info Follows:
Currently processing none
Maintenance Mode Set to None (not in mtc mode)
Number of signaling protocol errors are 0
Impedance is set to 600r Ohm
Station name None, Station number None

Caller ID Info Follows:
Standard ETSI DTMF
Output attenuation is set to 14 dB
Caller ID is transmitted after 1 ring(s)
Translation profile (Incoming):
Translation profile (Outgoing):

Voice card specific Info Follows:
Signal Type is loopStart
Ring Frequency is 25 Hz
Hook Status is On Hook
Ring Active Status is inactive
Ring Ground Status is inactive
Tip Ground Status is inactive
Digit Duration Timing is set to 100 ms
InterDigit Duration Timing is set to 100 ms
Hookflash-in Timing is set to max=800 ms, min=0 ms
Hookflash-out Timing is set to 400 ms
No disconnect acknowledge
Ring Cadence is defined by CPTone Selection
Ring Cadence are [10 40] * 100 msec
Ringer Equivalence Number is set to 1
################################################################################

#show running-config
Building configuration...

Current configuration : 3671 bytes
!
version 12.4
no service pad
service timestamps debug datetime msec
service timestamps log datetime msec
service password-encryption
!
hostname TBR-VG20401-RXX_ITABORAI
!
boot-start-marker
boot-end-marker
!
logging message-counter syslog
!
no aaa new-model
ip source-route
!
!
!
!
ip cef
no ipv6 cef
!
stcapp ccm-group 1
stcapp
!
stcapp feature access-code
!
stcapp feature speed-dial
!
!
!
stcapp call-control mode feature
!
!
!
!
voice service voip
fax protocol pass-through g711alaw
!
!
!
!
!
!
!
!
!
!
!
!
!
voice-card 0
dsp services dspfarm
!
username admin privilege 15 password 7 111D1C070513
archive
log config
  hidekeys
!
!
!
!
!
interface FastEthernet0/0
description ## Interface VOIP ##
ip address 10.23.31.4 255.255.255.0
duplex auto
speed auto
!
interface FastEthernet0/1
no ip address
shutdown
duplex auto
speed auto
!
ip default-gateway 10.23.31.1
ip forward-protocol nd
ip route 0.0.0.0 0.0.0.0 10.23.31.1
!
no ip http server
!
!
!
control-plane
!
!
!
voice-port 0/0
compand-type a-law
cptone BR
timeouts interdigit 3
timeouts call-disconnect 0
timeouts ringing infinity
timing hookflash-in 800 0
description FAX 1420
bearer-cap Speech
caller-id enable
!
voice-port 0/1
compand-type a-law
cptone BR
timeouts interdigit 3
timeouts call-disconnect 0
timeouts ringing infinity
timing hookflash-in 800 0
description FAX 1421
bearer-cap Speech
caller-id enable
!
voice-port 0/2
compand-type a-law
cptone BR
timeouts interdigit 3
timeouts call-disconnect 0
timeouts ringing infinity
timing hookflash-in 800 0
description FAX 1422
bearer-cap Speech
caller-id enable
!
voice-port 0/3
compand-type a-law
cptone BR
timeouts interdigit 3
timeouts call-disconnect 0
timeouts ringing infinity
timing hookflash-in 800 0
description FAX 1423
bearer-cap Speech
caller-id enable
!
ccm-manager fax protocol cisco
ccm-manager sccp local FastEthernet0/0
!
!
sccp local FastEthernet0/0
sccp ccm 10.23.44.3 identifier 2 version 7.0
sccp ccm 10.23.44.4 identifier 1 version 7.0
sccp
!
sccp ccm group 1
description ## UCM BR FARM ##
associate ccm 1 priority 1
associate ccm 2 priority 2
associate profile 1 register CFB_BR-ITABORAI
!
dspfarm profile 1 conference
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
shutdown
!
!
dial-peer voice 200 pots
service stcapp
port 0/0
!
dial-peer voice 202 pots
service stcapp
port 0/2
!
dial-peer voice 201 pots
service stcapp
port 0/1
!
dial-peer voice 203 pots
service stcapp
port 0/3
!
!
banner motd ^CCC

[Marcelo Bergamini]

Hi Aokanlawon!!

This device doesn´t suport conference bridge profile. It work´s only with conference bridge software on CCM or other device with DSPs avaiable.

[Ayodeji Okanlawon]

Oh I didnt realise this was a VG gateway device...

Ok. can you send a picture of the MRGL and MRG assiged to the sccp device in cucm

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[Sadaseeven Saminaden]

Marcelo.

I understand your point.

You have pasted config for the VG204.

Can you please share the output of the router that acts as your hardware conference bridge?

[Marcelo Bergamini]

The router that acts how hardware its ok, because IP phones work´s fine.

The analog phone doesn´t need only hardware conference bridge, because it can wok´s with CCM´ resource. Conference bridge sofware. But it doesn´t work.

[Ayodeji Okanlawon]

Can you send CUCM traces? Do you know how to enable cucm traces?

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[Marcelo Bergamini]

I have some notion about it, but I need to know how to enable properly.

You can pass me the way?

[Ayodeji Okanlawon · Accepted solution]

Use this link to enable the traces..

http://www.cisco.com/en/US/products/sw/voicesw/ps556/products_tech_note09186a0080094e89.shtml#calm

Then use this link to learn how to collect the traces abd send here..

https://supportforums.cisco.com/docs/DOC-9283

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[Marcelo Bergamini]

OK. I picked up the traces. They must be collected during the attempts to turn the conference?

  Here are the pictures requested. I used two types, one using a hardware gateway on the same site and another tantando only use resources CCM.

I have used the follows MRGL and MRG:

[Ayodeji Okanlawon · Accepted solution]

Yes enable the traces and then do your conference test..once you have finished collect thr traces and attach them here. Dont paste them

Please also send the calling and called numbers..or the extension of the phones involved

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[sumitkum]

To isolate the issue, Need some more information

1. the analog phones, IP Phones and PSTN line involved in the Confernce, all are in the same Region or not .

2. if make conf call between analog phones ( Region = default) connected to VG204 only, is it working fine.

3. what Region assing to analog phones, IP Phones, and PSTN line thru Device Pool.

[Marcelo Bergamini]

Hi Sumit!

1 - The involved in the Conference,are in the same region and use G.711.

2 - Its isn´t possivel because there are only one analog phone in this VG204. The others ports are FAX.

3 - Tha analog phones, IP Phones, and PSTN line use the same Region inside Device pool, but to achieve CCM (HQ), where there are conference resources they use, the system default interregion, G.729.

[Ayodeji Okanlawon]

Marcelo,

If the region between the Analogue phones and the CFB is G729, then it wont work. i thought you said its all G711. CUCM Software conference bridge does not support G729. Unless you have xcoder invoked. So configure the analogue phones to use G711 to the CFB or use hardware CFB

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[Marcelo Bergamini]

Actually I configured a specific Region for analog phones reach the CCM (HQ) using G.711.

I did this only for the analog phone, which starts the conference, but did not for the rest of the IP phones.

Sorry for the confusion!

[Ayodeji Okanlawon]

Can you post the picture of the region setting between the analogue phone and the CFB device.

Where are the traces?

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