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Chopping Audio From PSTN to Local IP Phone

Hi all, good day.

I have a problem with the PSTN,

In one of our branch offices we are experience some audio chopping, the person who is calling form PSTN can hear de audio good, but the caller VoIP person suddenly starts to hear no audio like form 1 to 2 seconds, but PSTN person can hear him all the time.

Here is how PSTN is connected.

--> E1 ds0 group --> Ruter --> Swtich --> VoIP Phone.

That would be the RTP flow, actually our branch office is connected to a remote call manager cluster for calling centralization  control, but that offices has its own router, its own PSTN E1 port, and its own DSP resources.

Here is some configuration.

controller E1 0/1/0

framing NO-CRC4

line-termination 75-ohm

ds0-group 0 timeslots 1-15,17-22 type r2-digital r2-compelled

cas-custom 0

  country telmex

  category 2

  answer-signal group-b 1

  caller-digits 4

  groupa-callerid-end

!

voice-port 0/1/0:0

cptone MX

timeouts interdigit 2

!

dspfarm profile 2 transcode

codec g711ulaw

codec g711alaw

codec g729ar8

codec g729abr8

codec g729r8

codec g729br8

maximum sessions 4

associate application SCCP

!

dial-peer voice 2000 voip

preference 1

destination-pattern 80437....

modem passthrough nse codec g711ulaw

voice-class h323 1

session target ipv4:10.2.50.9

dtmf-relay h245-alphanumeric

codec g711ulaw

fax rate disable

no vad

!

dial-peer voice 2001 voip

preference 2

destination-pattern 80437....

modem passthrough nse codec g711ulaw

voice-class h323 1

session target ipv4:10.2.50.5

dtmf-relay h245-alphanumeric

codec g711ulaw

fax rate disable

no vad

!

dial-peer voice 1001 pots

destination-pattern 9T

direct-inward-dial

port 0/1/0:0

no register e164

!

#show dps

Dspfarm Profile Configuration

Profile ID = 2, Service = TRANSCODING, Resource ID = 1 

Profile Description : 

Profile Admin State : UP

Profile Operation State : ACTIVE

Application : SCCP   Status : ASSOCIATED

Resource Provider : FLEX_DSPRM   Status : UP

Number of Resource Configured : 4

Number of Resource Available : 4

Codec Configuration

Codec : g711ulaw, Maximum Packetization Period : 30

Codec : g711alaw, Maximum Packetization Period : 30

Codec : g729ar8, Maximum Packetization Period : 60

Codec : g729abr8, Maximum Packetization Period : 60

Codec : g729r8, Maximum Packetization Period : 60

Codec : g729br8, Maximum Packetization Period : 60

Dspfarm Profile Configuration

SLOT DSP VERSION  STATUS CHNL USE   TYPE   RSC_ID BRIDGE_ID PKTS_TXED PKTS_RXED

0    1   8.4.2    UP     N/A  FREE  xcode  1      -         -         -       

0    1   8.4.2    UP     N/A  FREE  xcode  1      -         -         -       

0    1   8.4.2    UP     N/A  FREE  xcode  1      -         -         -       

0    1   8.4.2    UP     N/A  FREE  xcode  1      -         -         -       

0x1F5E4   19D9 0x673EADB4 0/1/0:0.20 0/3:4   804377101 g711ulaw 1001/2000

#show ver

Cisco IOS Software, 3800 Software (C3825-SPSERVICESK9-M), Version 12.4(4)XC7, RELEASE SOFTWARE (fc1)

Technical Support: http://www.cisco.com/techsupport

Copyright (c) 1986-2007 by Cisco Systems, Inc.

Compiled Wed 01-Aug-07 02:10 by alnguyen

ROM: System Bootstrap, Version 12.4(13r)T11, RELEASE SOFTWARE (fc1)

gwhervpn uptime is 2 weeks, 4 days, 22 hours, 42 minutes

System returned to ROM by reload at 10:19:18 GMT-7 Sat Oct 20 2012

System restarted at 10:22:01 GMT-7 Sat Oct 20 2012

System image file is "flash:c3825-spservicesk9-mz.124-4.XC7.bin"

#show inv

NAME: "Two port E1 voice interface daughtercard on Slot 0 SubSlot 1", DESCR: "Two port E1 voice interface daughtercard"

PID: VWIC-2MFT-E1=     , VID: 1.0, SN: 32867702  

NAME: "PVDMII DSP SIMM with four DSPs on Slot 0 SubSlot 4", DESCR: "PVDMII DSP SIMM with four DSPs"

PID: PVDM2-64          , VID: NA  , SN: FOC085028LJ

On some users we have trying change the phone for 7911 to 6921, didnt worked.

We have changed te call manager subscriber who controlls all the phones, didn't worked either.

Could you help me on this?

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