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@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@ui.tagnet.cc)}" /> <action application="hangup"/> </condition> </extension>
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005