[Freeswitch-users] DTMF problems

[Boris Kovalenko]

Hello!

     Kerem, are You talking about input gain parameter? AFAIK it is 0db 
by default. Would You recomend to set it lower?
ISDN 1/0:D - 1/0:D
  Type of VoicePort is ISDN
  Operation State is DORMANT
  Administrative State is UP
  No Interface Down Failure
  Description is Link to ATSE-41
  Noise Regeneration is disabled
  Non Linear Processing is disabled
  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 0 dB
  Echo Cancellation is enabled
  Echo Cancellation NLP mute is disabled
  Echo Cancellation NLP threshold is -21 dB
  Echo Cancel Coverage is set to 32 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 250 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 10 s
  Interdigit Time Out is set to 10 s
  Call Disconnect Time Out is set to 60 s
  Ringing Time Out is set to 180 s
  Wait Release Time Out is set to 30 s
  Spe country is russia
  Region Tone is set for RU
  Station name None, Station number None
  Translation profile (Incoming):
  Translation profile (Outgoing):

> Hi Boris,
>
> I have had same kind of problems with PRI circuits between Channel Banks
> and VoIP Servers due to static on the POTS line and/or too high gain
> settings on the PRI cicuit. Try playing with the gain settings on the
> Cisco 5350, maybe that can help.
>
> Regards,
> Kerem
>
>
>
> On Tue, 24 May 2011 15:11:08 +0300, Boris Kovalenko<boris at tagnet.ru>
> wrote:
>
>> Hello!
>>
>>       My network enviroment is: PRI --- Cisco 5350 --- FreeSwitch
>> (Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)) ---
>> AudioCodes MP-118FXS. There is an IVR on FreeSwitch where calling party
>> may press a key for personal client line or wait for a secretary.
>> Sometimes (as client said) the call to line 5 is placed to line 3 (as I
>> undestand DTMF 5 is recognized as 3). How may I detect and solve the
>> problem? Configurations below:
>> Cisco 5350:
>> dial-peer voice 4000 voip
>>    description OUTBOUND-CALLS-FROM-USI
>>    destination-pattern 50000#.T
>>    voice-class codec 1
>>    session protocol sipv2
>>    session target ipv4:82.193.138.187:5060
>>    dtmf-relay rtp-nte
>>    fax-relay ecm disable
>>    fax rate 9600
>>    fax protocol t38 ls-redundancy 5 hs-redundancy 0 fallback pass-through
>> g711alaw
>>    no vad
>>
>> FreeSwitch:
>> profile:
>> <param name="dtmf-duration" value="2000"/>
>> <!--<param name="dtmf-type" value="rfc2833"/>  -->  (yes, it is commented)
>>
>> context:
>> <extension name="ext_001">
>> <condition field="destination_number" expression="^(73435230100)$">
>> <action application="set" data="v_tagnet_ats_dstport=10005"/>
>> <action application="set" data="v_ats_dstport=10005"/>
>> <action application="set" data="import=v_tagnet_dstport,v_ats_dstpor
>> t"/>
>> <!--<action application="set" data="import=v_tagnet_ats_dstport"/>
>> -->
>> <action application="set" data="hangup_after_bridge=true"/>
>> <!--<action application="set" data="continue_on_fail=USER_NOT_REGIS
>> TERED"/>  -->
>> <action application="set" data="ringback=$${ru-ring}"/>
>> <action application="javascript" data="${scripts_dir}/common/set_dpe.js
>> B ui.tagnet.cc 3 3"/>
>> <action application="answer"/>
>> <action application="sleep" data="1000"/>
>> <action application="play_and_get_digits" data="1 1 1 5000 #
>> /opt/fs/sounds/ntmjui/welcome.wav silence_stream://250 digits \d* 3000"/>
>> <action application="execute_extension" data="ext_call_internal"/>
>> <action application="bridge" data="${sofia_contact(101 at ui.tagnet.cc)}" />
>> <action application="hangup"/>
>> </condition>
>> </extension>
>>
>>
>


-- 
С уважением,
   Борис Коваленко
   ЗАО "Тагнет"
   тел. +7 (3435) 230001
   факс +7 (3435) 230005