DTMF

Similar Questions

• Get the right order of tones for DTMF decoder

  I'm trying to decoder stream DTMF signal (contains mutiple tones, delay, noise), however, I have the problem to separate each tone of the stream signal in order to enter the Goertzel for later analysis.

  The phone number is 534-343-3436. To the decoder input, I don't know what's the phone number again. However, I want to separate 5 then 3 then 4 then 3... your if I put analysis individual tone is what key corresponding to this phone number.

  Anyway I can archieve in LabView?

  There should always be an interval of no signal between pairs of tone - if not, you can not detect repeating values. For phone number 534-343-3333 you can not just look for a frequency change. Your signaling protocol must specify the minimum length of a pair of tone to be considered a valid signal and a minimum without your being a separation valid between the numbers. Maximum number of hours is useful, but not as important, except to determine how long the system will wait for a valid number.  Time settings may depend on the acceptable minimum signal to noise ratio.

  The algorithm would be something like this: amplitude discrimination allows to find gaps between pairs of tone. There is no need to be too good - just find approximate segments. It will use timing specifications for set time periods to consider. Within each segment threshold use the FFT to identify the pair of tone. You can also use a FFT on the gap segments to check that they do not contain a pair of tone.

  Also consider various errors. On tampons in your Western Electric simultaneously pressing two or more keys in a row or a column results in a single tone. I have not watched the DTMF generators for a few years so I don't know how how some of the IC-based generators to handle this situation.

  Lynn

• Break DTMF

  I saw in the posts on this forum the two "."  and ',' (period and comma) which means a break in the sending of DTMF tones.  I found no "Official" definition Is this one?

  How I she applied, is the API that is just taking a break?

  PhoneCall.sendDTMFTones does not support pauses (commas or periods).  Your application will need to wait/sleep between tones if you want to have a delay between the tones.

• DTMF in SIP Trunk problem

  Hello

  I have a problem in case of detection of the DTMF

  We have a SIP of the ITSP Trunk and everything is ok except DTMF.

  The sip trunk is between ITSP and router 3945

  ITSP <->3945 <->CUCM 10.5

  I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs

  ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us

  16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
  Received:
  INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0
  Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8
  Call ID: [email protected]/ * /.
  From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40
  To: sip: [email protected] / * /; user = phone >
  CSeq: 1 INVITE
  Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, see
  Max-Forwards: 69
  Supported: 100rel, timer
  User-Agent: Huawei SoftX3000 V300R010
  Session time-out: 300
  Min - SE: 90
  Contact: sip: [email protected] / * /: 5060; user = phone >
  Content-Length: 374
  Content-Type: application/sdp
  v = 0
  o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34
  s = call Sip
  c = IN IP4 10.105.40.34
  t = 0 0
  m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102
  a = rtpmap:8 PCMA/8000
  a = rtpmap:0 PCMU/8000
  a G729/8000 rtpmap:18 =
  a = rtpmap:4 G723/8000
  a = rtpmap:2 G726-32/8000
  a = rtpmap:98 G726-40/8000
  a = rtpmap:99 G726-32/8000
  a = rtpmap:102 G726-24/8000
  a = ptime:20
  a = fmtp:18 annex b = No.
  It is a message to guest (with sdp) of ITSP
  As you can see the line with red color must have a code with number of 101 but rather a code with number of 18
  In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. '
  It's my router config:
  voip phone service
  No IP trust to authenticate
  allow connections h323 to SIP
  allow connections sip h323
  allow sip to sip connections
  SIP
  interface FastEthernet0/0/1 source control binding
  bind media source interface FastEthernet0/0/1
  min - to 300 session expires-300
  !

  Dial-peer voice 2 voip---> router CUCM and vice versa
  translation-profile outgoing toos
  destination-model 42584...
  session protocol sipv2
  session target ipv4:10.20.30.70
  Codec g711ulaw
  DTMF-relay rtp - nte
  !
  VoIP voice 10 Dial - peer---> router for ITSP and vice versa
  destination-model. T
  session protocol sipv2
  session target ipv4:10.105.40.34
  incoming called-number. T
  DTMF-relay rtp - nte
  Codec g711ulaw
  I have configured cucm with a sip section to my favorite router with active PSG and RFC2833
  BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOING
  I even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not work
  I change the codec but does not solve the problem
  There is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)
  Please give me a solution to solve the problem between Cisco 3945 and ITSP
  Concerning

  It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.

• [CME 9.1] Phone IP 3905 fails to send DTMF tones to the external network.

  Hi all!

  We have three types of SIP phones in our CME 9.1: 3905, 6941 and 8941. All phones except 3905 send DTMF tones successfully through our operator. Here is "debug voice ccapi detail" what "206" (ip phone 3905) extension 93274343 external phone calls and trying to send numbers. Config is to be attached. Also, a part of config here:

  voip phone service

  list of approved IP addresses

  IPv4 10.0.0.0

  h323 connections allow h323

  allow connections h323 to SIP

  allow connections sip h323

  allow sip to sip connections

  service additional h450.12

  no additional service moved temporarily sip

  no service additional sip refer

  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none

  H323

  SIP

  Registration Server expires 120 min 60 max

  !

  voice class codec 1

  g711ulaw codec preference 1

  g711alaw preferably 2 codec

  !

  !

  Global voice registry

  FMC of fashion

  source-address 10.10.0.41 port 5060

  3 timeouts interdigit

  Max - dn 100

  Max-pool 80

  load 8961 sip8961.9 - 2 - 2 SR 1-9

  authenticate the registry

  authenticate the defagroup.com Kingdom

  time format 24

  date format D/M/Y

  Flash TFTP-path:

  create the profile synchronization 0001013651736503

  Hi all!

  We have three types of SIP phones in our CME 9.1: 3905, 6941 and 8941. All phones except 3905 send DTMF tones successfully through our operator. Here is "debug voice ccapi detail" what '206' extension 93274343 external phone calls and trying to send numbers. Config is to be attached. Also, a part of config here:

  voip phone service

  list of approved IP addresses

  IPv4 10.0.0.0

  h323 connections allow h323

  allow connections h323 to SIP

  allow connections sip h323

  allow sip to sip connections

  service additional h450.12

  no additional service moved temporarily sip

  no service additional sip refer

  Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none

  H323

  SIP

  Registration Server expires 120 min 60 max

  !

  voice class codec 1

  g711ulaw codec preference 1

  g711alaw preferably 2 codec

  !

  !

  Global voice registry

  FMC of fashion

  source-address 10.10.0.41 port 5060

  3 timeouts interdigit

  Max - dn 100

  Max-pool 80

  load 8961 sip8961.9 - 2 - 2 SR 1-9

  authenticate the registry

  authenticate the defagroup.com Kingdom

  time format 24

  date format D/M/Y

  Flash TFTP-path:

  create the profile synchronization 0001013651736503

  Register of voice dn 6

  number 204

  call-forward noan 201 timeout 20 b2bua

  Register of voice model 1

  function key 1 Redial

  function key 2 Cfwdall

  function key 3 Hold

  function key 5 Trnsfer

  function key 6 DND

  !

  voice dialing plan registry 1

  type of 7940-7960-others

  model 1...

  2 9810 model *.

  model 3 9...

  Model 4 98...

  Register of voice pool 8

  Mac ID 64D8.14A5.01B4

  type of 3905

  Number 1 dn 8

  numbering plan 1

  DTMF-relay rtp - nte

  codec voice-class 1

  206 206 username password

  No vad

  voice POTS dial-peer 1

  translation-profile out in the city

  destination-model 9 t

  port 0/0/0:15

  Retail ccapi voice 'debug' is here:

  phone #.

  * Dec 27 16:45:22.095: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: //-1/xxxxxxxxxxxx/CCAPI/cc_api_update_interface_cac_resource:

  Hwidb = GigabitEthernet0/0, bandwidth = 80, Call Id = 817

  * Dec 27 16:45:22.095: //-1/xxxxxxxxxxxx/CCAPI/cc_api_update_interface_cac_resource:

  Call total count = 0, call Voip Count = 0, Count MMoip call = 0 x 0, bandwidth = 80

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:22.095: //-1/A3B9036582FD/CCAPI/cc_api_call_setup_ind_common:

  Interface type = 0, Protocol = 3

  * Dec 27 16:45:22.095: //-1/A3B9036582FD/CCAPI/ccCheckClipClir:

  Part number is provided by the user

  * Dec 27 16:45:22.095: //-1/A3B9036582FD/CCAPI/cc_api_call_setup_ind_common:

  After checking the number translation:

  Number = 206 (TON = unknown, NPI = unknown, not projected = screening, presentation = authorized),

  Called number = 3 (TONNE = unknown, NPI = unknown)

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

  Total Call Count = 0, entry calls (call count On = FAKE incoming call = TRUE)

  * Dec 27 16:45:22.095: / / 817/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

  Total Call Count = 1

  * Dec 27 16:45:22.095: / / 817, A3B9036582FD, CCAPI, cc_insert_guid_pod_entry:

  Incoming = TRUE, Call Id = 817

  * Dec 27 16:45:22.095: / / 817, A3B9036582FD, CCAPI, cc_incr_if_call_volume:

  10.10.0.126 = remote IP address, Hwidb = GigabitEthernet0/0

  phone #.

  * Dec 27 16:45:22.095: / / 817, A3B9036582FD, CCAPI, cc_incr_if_call_volume:

  Call count total = 1 call Voip Count = 1, Call MMoip Count = 0

  * Dec 27 16:45:22.099: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:

  Corresponding settings; Called number = 3, call transfer Consult Id =

  * Dec 27 16:45:22.099: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:

  No matching node

  * Dec 27 16:45:22.099: / / 817, A3B9036582FD, CCAPI, cc_api_set_transfer_info:

  Call transfer Reset

  * Dec 27 16:45:22.443: / / 817, A3B9036582FD, CCAPI, ccCallDisconnect:

  Start calling accounting;

  Call Entry (Incoming = TRUE)

  * Dec 27 16:45:22.443: / / 817, A3B9036582FD, CCAPI, ccCallDisconnect:

  Value = 28, entry calls (disconnect the Cause = 0)

  * Dec 27 16:45:22.443: / / 817, A3B9036582FD, CCAPI, cc_api_update_interface_cac_resource:

  Hwidb = GigabitEthernet0/0, bandwidth =-80, Call Id = 817

  * Dec 27 16:45:22.443: / / 817, A3B9036582FD, CCAPI, cc_api_update_interface_cac_resource:

  Call count total = 1 call Voip Count = 1, Call MMoip Count = 0x0, bandwidth = 0

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, cc_decr_if_call_volume:

  10.10.0.126 = remote IP address, Hwidb = GigabitEthernet0/0

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, cc_decr_if_call_volume:

  Call total count = 0, call Voip Count = 0, Call MMoip Count = 0

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, cc_delete_guid_pod_entry:

  Incoming = TRUE

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, cc_delete_call_entry:

  Total Call Count = 1, entry calls (call count On = FAKE incoming call = TRUE)

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, cc_delete_call_entry:

  Total Call Count = 0

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, cc_delete_call_entry:

  Removal of profileTable [0x336F00D4]

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, ccCallGetVoipFlag:

  Mask of data bits = 0 x 2, Call Id = 817

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, ccCallGetVoipFlag:

  Flag = FALSE

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, ccCallSetVoipFlag:

  telephony #Flag = FALSE, data bits mask = 0 x 2, call Id = 817

  * Dec 27 16:45:22.559: / / 817, A3B9036582FD, CCAPI, ccCallSetVoipFlag:

  Call the entry (Voip AAA Flags = 0x0)

  * Dec 27 16:45:22.559: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  phone #.

  * Dec 27 16:45:28.995: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: //-1/xxxxxxxxxxxx/CCAPI/cc_api_update_interface_cac_resource:

  Hwidb = GigabitEthernet0/0, bandwidth = 80, Call Id = 818

  * Dec 27 16:45:28.995: //-1/xxxxxxxxxxxx/CCAPI/cc_api_update_interface_cac_resource:

  Call total count = 0, call Voip Count = 0, Count MMoip call = 0 x 0, bandwidth = 80

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:28.995: //-1/A7D5DC978303/CCAPI/cc_api_call_setup_ind_common:

  Interface type = 0, Protocol = 3

  * Dec 27 16:45:28.995: //-1/A7D5DC978303/CCAPI/ccCheckClipClir:

  Part number is provided by the user

  * Dec 27 16:45:28.995: //-1/A7D5DC978303/CCAPI/cc_api_call_setup_ind_common:

  After checking the number translation:

  Number = 206 (TON = unknown, NPI = unknown, not projected = screening, presentation = authorized),

  Called number = 9 (TON = unknown, NPI = unknown)

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

  Total Call Count = 0, entry calls (call count On = FAKE incoming call = TRUE)

  * Dec 27 16:45:28.995: / / 818/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

  Total Call Count = 1

  * Dec 27 16:45:28.995: / / 818/A7D5DC978303/CCAPI/cc_insert_guid_pod_entry:

  Incoming = TRUE, Call Id = 818

  * Dec 27 16:45:28.999: / / 818/A7D5DC978303/CCAPI/cc_incr_if_call_volume:

  10.10.0.126 = remote IP address, Hwidb = GigabitEthernet0/0

  phone #.

  * Dec 27 16:45:28.999: / / 818/A7D5DC978303/CCAPI/cc_incr_if_call_volume:

  Call count total = 1 call Voip Count = 1, Call MMoip Count = 0

  * Dec 27 16:45:28.999: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:

  Corresponding settings; Called number = 9, call transfer Consult Id =

  * Dec 27 16:45:28.999: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_registration_lookup:

  No matching node

  * Dec 27 16:45:28.999: / / 818/A7D5DC978303/CCAPI/cc_api_set_transfer_info:

  Call transfer Reset

  phone #.

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:

  VALIDATION MLPP INFO:-ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_is_precedence_mlpp_info:

  Priority no more than Routine

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_copy_mlpp_info:

  Before COPYING - CBC MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_copy_mlpp_info:

  DEST MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_copy_mlpp_info:

  AFTER COPY - CBC MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)] DEST MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [-1 (PRECEDENCE_LEVEL_NONE)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_copy_mlpp_info:

  Before COPYING - CBC MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_copy_mlpp_info:

  DEST MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [-1 (PRECEDENCE_LEVEL_NONE)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_copy_mlpp_info:

  AFTER COPY - CBC MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)] DEST MLPP INFO: ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [-1 (PRECEDENCE_LEVEL_NONE)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:

  VALIDATION MLPP INFO:-ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [0 (INTERNAL_PRECEDENCE_0)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_is_precedence_mlpp_info:

  Priority no more than Routine

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_validate_mlpp_info:

  VALIDATION MLPP INFO:-ServiceDomain: [no (0)] DomainIdentifier: [000000] PrecedenceLevel: [-1 (PRECEDENCE_LEVEL_NONE)] NormalizedPrecedence: [-1 (PRECEDENCE_LEVEL_NONE)]

  * Dec 27 16:45:34.827: //-1/xxxxxxxxxxxx/CCAPI/cc_is_precedence_mlpp_info:

  Priority no more than Routine

  * Dec 27 16:45:34.827: / / 818/A7D5DC978303/CCAPI/ccCheckClipClir:

  Part number is provided by the user

  * Dec 27 16:45:34.827: / / 819/xxxxxxxxxxxx/CCAPI/cc_insert_call_entry:

  Total Call Count = 1, entry calls (call count On = FAKE incoming call = FALSE)

  * Dec 27 16:45:34.827: / / 818/A7D5DC978303/CCAPI/cc_peer_bind:

  Link = TRUE, Id = 818, bound Call Id = 819 Call Binder

  * Dec 27 16:45:34.827: / / 819, A7D5DC978303, CCAPI, cc_insert_guid_pod_entry:

  Incoming = FALSE, Call Id = 819

  phone #.

  * Dec 27 16:45:34.827: / / 819, A7D5DC978303, CCAPI, cc_set_voice_port_value:

  CC_IF_TELEPHONY: Echo = 0, = 0 Playout

  * Dec 27 16:45:34.831: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:34.831: / / 818/A7D5DC978303/CCAPI/ccCallGetContext:

  Context = 0x33C6F520, Id = 818 Call

  * Dec 27 16:45:34.831: //-1/xxxxxxxxxxxx/CCAPI/cc_set_outpulsed_digits:

  set 3274343 = outpulsed_dialstring

  * Dec 27 16:45:35.399: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:35.399: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:45:35.403: / / 818/A7D5DC978303/CCAPI/cc_peer_bind:

  phone #Bind = TRUE, Binder Call Id = 818, bound Call Id = 819

  * Dec 27 16:45:35.559: / / 818/A7D5DC978303/CCAPI/cc_peer_bind:

  Link = TRUE, Id = 818, bound Call Id = 819 Call Binder

  phone #.

  * 16:45:45.199 Dec 27: % LINEPROTO-5-UPDOWN: Line protocol on Interface Serial0/0/0:14, status changed to

  phone #.

  * Dec 27 16:46:02.215: / / 818/A7D5DC978303/CCAPI/ccCallDisconnect:

  Start calling accounting;

  Call Entry (Incoming = TRUE)

  * Dec 27 16:46:02.215: / / 818/A7D5DC978303/CCAPI/ccCallDisconnect:

  Value = 16, entry calls (disconnect the Cause = 16)

  * Dec 27 16:46:02.215: / / 818/A7D5DC978303/CCAPI/ccCallDisconnect:

  Call the entry (disconnect the Cause = 16)

  * Dec 27 16:46:02.215: / / 819, A7D5DC978303, CCAPI, ccCallDisconnect:

  Start calling accounting;

  Call Entry (Incoming = FALSE)

  * Dec 27 16:46:02.215: / / 819, A7D5DC978303, CCAPI, ccCallDisconnect:

  Value = 16, entry calls (disconnect the Cause = 0)

  * Dec 27 16:46:02.215: / / 818/A7D5DC978303/CCAPI/cc_api_update_interface_cac_resource:

  Hwidb = GigabitEthernet0/0, bandwidth =-80, Call Id = 818

  * Dec 27 16:46:02.215: / / 818/A7D5DC978303/CCAPI/cc_api_update_interface_cac_resource:

  Call count total = 1 call Voip Count = 1, Call MMoip Count = 0x0, bandwidth = 0

  * Dec 27 16:46:02.227: //-1/A7D5DC978303/CCAPI/g113_calculate_impairment:

  (delay = 79 (ms), loss = 0%), Qi = 0 participants Io = 0 = 0 = 2 Ie = - 1 Itot = 1 DLI

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/cc_decr_if_call_volume:

  10.10.0.126 = remote IP address, Hwidb = GigabitEthernet0/0

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/cc_decr_if_call_volume:

  Call total count = 0, call Voip Count = 0, Call MMoip Count = 0

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/cc_delete_guid_pod_entry:

  Incoming = TRUE

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/cc_delete_call_entry:

  Total Call Count = 1, entry calls (call count On = FAKE incoming call = TRUE)

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/cc_delete_call_entry:

  Total Call Count = 0

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/cc_delete_call_entry:

  Removal of profileTable [0x336EFA6C]

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/ccCallGetVoipFlag:

  Mask of data bits = 0 x 2, Call Id = 818

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/ccCallGetVoipFlag:

  Flag = FALSE

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/ccCallSetVoipFlag:

  telephony #Flag = FALSE, data bits mask = 0 x 2, call Id = 818

  * Dec 27 16:46:02.227: / / 818/A7D5DC978303/CCAPI/ccCallSetVoipFlag:

  Call the entry (Voip AAA Flags = 0x0)

  * Dec 27 16:46:02.227: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, cc_delete_guid_pod_entry:

  Incoming = FALSE

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, cc_delete_call_entry:

  Total Call Count = 0, entry calls (call count On = FAKE incoming call = FALSE)

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, cc_delete_call_entry:

  Removal of profileTable [0x336E75C4]

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, ccCallGetVoipFlag:

  Mask of data bits = 0 x 2, Call Id = 819

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, ccCallGetVoipFlag:

  Flag = FALSE

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, ccCallSetVoipFlag:

  Flag = FALSE, data bits mask = 0 x 2, Call Id = 819

  * Dec 27 16:46:02.275: / / 819, A7D5DC978303, CCAPI, ccCallSetVoipFlag:

  Call the entry (Voip AAA Flags = 0x0)

  * Dec 27 16:46:02.275: / / 0, xxxxxxxxxxxx, CCAPI, cc_get_call_entry:

  Entry call is not found

  phone #unde all

  The CME version is 9.1

  Version of the IOS is 15.3 (1) T (also tested in 15.2 - the same issue)

  All ip phones have the last firware (9-2-2-0), also I have already spent some of them (9-2-1-0) - the problem persists.

  Hello

  I think the best is to capture a trace of sniffer on the phone directly, we will be able to see if the phone itself sends the numbers.

  You can also enable some debugs:

  Deb voice ccapi inout

  messages ccsip deb

  Media ccsip deb

  That should be enough to isolate the problem

  --
  Jorge Armijo

  Do not forget to rate helpful responses and identify useful or correct answers.

• SPA303 DTMF not by manual settings

  DTMF for applications call does not work on my SPA303. The receiving computer cant 'hear' tones.

  Manual line indicates that it there two options - In-Band and Out-of-Band (RFC-2833) and goes on recommend out of band. But the config utility has 6 options

  1. In the band
  2. AVT
  3. INFO
  4. Auto (default)
  5. In the Strip more info
  6. AVT + INFO

  At least the manual should reflect the utility - however, my questions are;

  1. Assuming that WRN = Out of Band, select with or without INFO (whatever it is)?
  2. DTMF Tx Volume for AVT package: 0 (default) - manual does not mention. Is it OK?
  3. DTMF AVT package interval: 0 (default) - manual does not mention. Is it OK?

  FYI: favorite Codec: G711u (default)

  I have an a noob and barely understand all of this. Made manual and research forum before posting.

  Thank you

  Neal

  There is an in-band and two out-of-band method (AVT and INFORMATION). Also, it is possible to send the DTMF using both methods at the same time. It translates into 6 options you mentioned.

  Now your questions:

  1. you prefer method supported by your voice provider. 'auto' works correctly in most cases.

  2 is documented in Appendix A of the Administrator's guide. Don't miss related problems of compatibility of the bridge. Ask decent value for its VoIP service provider gateway. Try a nonzero value if no information is avaiable.

  3. This is undocumented option. Keep the default unless instructed to change.

  Codec used - it is beeter to avoid if possible the transcoding. G711u is preferred in North America, while G711a is used in Europe. Use codec preferred in the country of the voip provider, to which you are connected.

• When the video parameters SX, MX and CSeries will be a keyboard of DTMF tones on call control page?

  Currently, the page of call control on the web interface of the Cisco video endpoints doesn't have a way to enter DTMF tones remotely.   It doesn't seem to be a keyboard of DTMF tones available to enter these tones remotely.   Does anyone know if Cisco will be adding this soon?

  You can enter DTMF tones using the web interface under Setup > API.

  xCommand DTMFSend CallId: DTMFString:

  • CallId: The CallID returned by operating the xCommand command Dial. During the call, you can run the xStatus calling command to see the CallId.
  • DTMFString: Enter the DTMF string.

  Example: xCommand DTMFSend CallId:2 DTMFString:1234

  Best place to find updates to come or possible would be to contact your Cisco account manager, if nothing is being developed or looked at the current time, they can create a demand for improvement for you.  However, the C series and the earlier boundaries of MX series points are already end of sale, and TC software they are running is in maintenance mode, in order to get the new features added to these devices special is unlikey.  Although there is a chance to get it added to the current range of endpoints SX and MX Series, just need to contact your Cisco AM and see what they can do to make your request.

• DTMF tone speed

  Had a look on the forums but couldn't find a solution.

  I guess there is no way to speed up the rate at which a DTMF tone is sent after the other?

  No, if you press the keys during a call, dtmf tones are added to send it queue all the same, no difference in speed.

• Are other than 0-9 DTMF tones, #, * supported?

  According to the official specifications of DTMF, there are 16 DTMF tones:

  0-9, #, *, A, B, C, D

  Are all supported by BlackBerry when using PhoneCall.sendDTMFTones ()?

  A, B, C, D, are not mentioned in the documentation.

  lol the supported are in the API.

• BasicEditField DTMF. What filter to use?

  This is a quickie for the legends of the forum

  Filter which do you suggest me for a DTMF input field?

  Permit characters would be: a - z, 0-9, *, #.

  Anyone can find the right combination of flags of filter for something like that?

  Thank you!

  a to z? as dtmf?

  There is no predefined filter for this. You can work with keydown or keychar, depending on what exactly you want to do.

• How to test DTMF to the unit

  How to test the sequience DTMF #X #2 in the unit. I have personal assistant on the system and when I send voive mail it says "I don't recognize that as a valid entry" unit is integrated with Domino.

  If you're component unit, pending so that he could respond and then entering DTMF? OK, so you don't need the leader ' # ' in your string to it. Assuming you get the opening greeting with your call, you can simply dial the extension of the user followed by #2 on turn the transfer rule.

  You can check what DTMF is entered and how the conversation flies with the Port status monitor tool that you will find in the depot of tools on your desktop - it must be reasonably obvious how the flow goes in looking at the output from that on a test call.

• WebEx h323 dtmf relay

  Hello

  I am facing a problem with endpoint h323 calling for webex.

  It connects to the right if has been previously logged by an organizer. When I try to connect as an organizer with the h.323 endpoint, it fails. The endpoint is autonomous, not registered in any guard.

  I can hear the tones generated in the endpoint, but webex does not seem to receive.

  So I would like to know what h323 dtmf relay mechanisms are supported in webex.

  Anyone know where I can find this info?

  Hello Ignacio,

  If you are referencing a CMR-Cloud here, for H.323 DTMF CMR-Cloud room would support h245-signal only.

  -Jonathan

• 'call send DTMF' of the CTS

  Anyone know the syntax of this CLI command without papers?  One can understand...

  Hello

  SYNTAX:

  call send DTMF {Arg0} {Arg1}

  DESCRIPTION:

  Arg0 - mandatory

  Caller ID for which send the DTMF signal

  Arg1 - mandatory

  DTMF to send signal

  Example, here you are...:

  1. start the call:

  Admin:call start 112284021490001

  2. once the call is set up check Call command ID:

  Status of the call admin:Show

  Call status

  Recorded in Cisco Unified Communications Manager: Yes

  Call connected: Yes

  Type of call: call Audio only call start time: 19 09:39:27 Feb 2014

  Duration (sec): 15 Direction: outgoing

  Local number: 112284031421006 remote number: 112284021490001

  Status: answered

  Security level: unsecured call Id: 3

  3 send DTMF command:

  Admin:call send DTMF 3 53921568

  Send dtmf signals... FACT

  Admin:

  Concerning

  Marek

• TC software for values letter DTMF

  Hello!

  You have a question that I can't find an answer documented for.  On the endpoints running a version of TC software, is it possible to get the endpoint to generate for A, B, C or D DTMF tones?

  My gut feeling is no, if this is because the graphic DTMF key does not have these buttons (only 0-9, *, #).  The reason to ask, that's what a customer trying to call in a bridge of AT & T who has one of these letters in the AXIS.

  Thank you!

  You can do this via the command line:

   xcommand DTMFSend DTMFString: A

  Alternatively, you can send the PIN all at once, IE if the PIN is 1A2B3C4D5 you can:

   xcommand DTMFSend DMTFString: 1A2B3C4D5

  Wayne
  --
  Remember the frequency responses and mark your question as answered as appropriate.

• Would like more information on: DTMF, TTY, battery, meter etc. tones Service.

  • DTMF-long or short?
  • Installation of the ATS?
  • The battery meter?
  • Flight power off mode, airplane Mode invites to start/stop?
  • Your power Service network?
  • Call your Drop/off voltage.

  No details on what they are or what they do.  Help.  Thank you.

  Oh, and ATS is a device for the blind hearing