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> -- Regards, Boris
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>
-- Kerem Erciyes - Sistem Danışmanı http://keremerciyes.com
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
Yes Boris, try setting it lower incrementally and see if it helps. There might be problems related to gain + echo on the line that confuses DTMF detection, so check if the line has echo as well. Also sometimes I have seen that the remote calling party has a POTS line with too much static it can confuse DTMF detection. On other VoIP systems we solved this problem by spying on the channel e.g. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be helpful in debugging the situation. On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko <boris@tagnet.ru> wrote:
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:
<snip>
Hello! Thank You Kerem, I'll try. The main problem is that there is a little percent of wrong DTMF and a very nervous customer. Also what if not to use dtmf-relay on Cisco at all? What kind of problems may I have? I use G711alaw inside a network. Some transit calls (ie Internet -> PSTN) may use G729.
Yes Boris, try setting it lower incrementally and see if it helps. There might be problems related to gain + echo on the line that confuses DTMF detection, so check if the line has echo as well. Also sometimes I have seen that the remote calling party has a POTS line with too much static it can confuse DTMF detection. On other VoIP systems we solved this problem by spying on the channel e.g. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be helpful in debugging the situation.
On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko<boris@tagnet.ru> wrote:
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:
<snip>
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Regards, Boris
No problem Boris. I never had much success with in-band DTMF but as M. Collins suggested in a separate thread there are some cases where you really need in-band detection (i.e. calls via GSM networks always send in-band DTMF), I suggest you try recording the calls and try to make sense of how much of the DTMF tone is being received, if there is stuttering due to echo cancellation or static on the line, etc... On Tue, 24 May 2011 16:57:15 +0300, Boris Kovalenko <boris@tagnet.ru> wrote:
Hello!
Thank You Kerem, I'll try. The main problem is that there is a little percent of wrong DTMF and a very nervous customer. Also what if not to use dtmf-relay on Cisco at all? What kind of problems may I have? I use G711alaw inside a network. Some transit calls (ie Internet -> PSTN) may use G729.
Yes Boris, try setting it lower incrementally and see if it helps. There might be problems related to gain + echo on the line that confuses DTMF detection, so check if the line has echo as well. Also sometimes I have seen that the remote calling party has a POTS line with too much static it can confuse DTMF detection. On other VoIP systems we solved this problem by spying on the channel e.g. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be helpful in debugging the situation.
On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko<boris@tagnet.ru> wrote:
<snip>
<snip> <snip>
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello! Hmm... we are talking about calls coming from PRI so (AFAIK) DTMF is always in-band, isn't? And Cisco is responsible to detect it and send out-of-band. IMHO bad call quality (e.g. call from GSM) is a good explanation.
No problem Boris. I never had much success with in-band DTMF but as M. Collins suggested in a separate thread there are some cases where you really need in-band detection (i.e. calls via GSM networks always send in-band DTMF), I suggest you try recording the calls and try to make sense of how much of the DTMF tone is being received, if there is stuttering due to echo cancellation or static on the line, etc...
On Tue, 24 May 2011 16:57:15 +0300, Boris Kovalenko<boris@tagnet.ru> wrote:
Hello!
Thank You Kerem, I'll try. The main problem is that there is a little percent of wrong DTMF and a very nervous customer. Also what if not to use dtmf-relay on Cisco at all? What kind of problems may I have? I use G711alaw inside a network. Some transit calls (ie Internet -> PSTN) may use G729.
Yes Boris, try setting it lower incrementally and see if it helps. There might be problems related to gain + echo on the line that confuses DTMF detection, so check if the line has echo as well. Also sometimes I have seen that the remote calling party has a POTS line with too much static it can confuse DTMF detection. On other VoIP systems we solved this problem by spying on the channel e.g. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop can be helpful in debugging the situation.
On Tue, 24 May 2011 16:10:03 +0300, Boris Kovalenko<boris@tagnet.ru> wrote:
<snip>
<snip> <snip>
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- С уважением, Борис Коваленко ЗАО "Тагнет" тел. +7 (3435) 230001 факс +7 (3435) 230005
participants (2)
-
Boris Kovalenko
-
Kerem Erciyes