Send dtmf or wherelse

Hello..

I have the next guy... I have to do an application what to do a procedure in a PBX.

at this moment the PBX have a Pentecost function that I can make a call and type a few keys send numers o a break... DTFM tones... and this Pentecost keys I program the PBX.

so, how do I do an application what some call to PBX in the background and send DTFM tones to make the program in the PBX?

Read this thread, it might be useful

http://supportforums.BlackBerry.com/T5/Java-development/sending-DTMFTones/m-p/557180#U557180

Tags: BlackBerry Developers

Similar Questions

[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.

'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

BlackBerry smartphones can not send (DTMF) digit code to open the Holy door - Bold 9790

I get a call from the door intercomm when a visitor arrives. I'm supposed to then "dial" while the call is open to open the door. It worked well for months, but now DO NOT SEND THE NUMBER SOUND.

There are some settings that must be changed, or has the spirit of darkness decended on the phone?

Help will be appreciated

Paul

Cold, rebooted and solved the problem. No need to answer more.

See you soon

Sending DTMF tones

Hello

I'm following the steps in the article available here

http://docs.BlackBerry.com/en/developers/deliverables/17953/Add_single_DTMF_tone_to_send_queue_56555...

but the Builder

new PhoneCall()

There is no

Indeed, the reference to the api class PhoneCall does not contain this constructor.

Where I'm wrong?

Thank you

the documentation is probably correct.

initiate a call is not blocking. This means that the following line of code runs before the appeal is made.
use a http://www.blackberry.com/developers/docs/6.0.0api/net/rim/blackberry/api/phone/PhoneListener.html to detect the connected call.

Obstructionisme!

I use a configuration with windows 7 (English, cant use Swedish, national characters in file names). Chrome, ripple, last simulatotrs, VMWare, latest version of the SDK. migrate a complex application from OS5 to OS10

-The simulator of VMWare will not work. Problem posted earlier. No soluition to date.

-Download the file does not work. People say its broken. Found no alternative.

-Trying to find how to make a phone call in the API documentation. Did not. Found an entry in the community to use

onclick = "BlackBerry.Launch.newPhoneCall ({'dialString': ' 5198887465', 'smartDialing': false})" / > "

-Got «undefined for blackberry»

-Updated config with:

Did not help.

Maybe try to make a few cc + extensions to solve problems? HM. Not a simple task. looking at the community effort on GIT.

Should I have the simulator of VMWare? Seem like it. Can not get how I could do a manula call on the simulator of the ripple.

Tired? Very. Frustrated? Yes

Any success? Yes

I can send a request to a Web site and get an answer using 'Workers', JSON objects and eventlisteners to

to run in the background. But which is basically just HTML5/Javascript, nothing specific BB. Workers + JSON is a very good platform to make a difference in the background.

Elwood: "it's 106 miles to Chicago, we have a full tank of gas, a packet of cigarattes, it's dark and we wear sunglasses." Jake: "it hit."

You will need to write a native extension to send DTMF tones. We do not support this in WebWorks. You can find an extension native example under the repo of community here:

https://github.com/BlackBerry/core-native-community-samples/tree/master/AudioLoopBackSample

Regarding the documentation for the phone, it is available here for WebWorks. Here's the new documents, we have set up a few weeks ago:

https://developer.BlackBerry.com/HTML5/documentation/phone.html

EventInjector works in the emulator but not on the phone.

Long time listener, first time caller...

(1) I use the version of the plugin or Eclipse JDE (. 64)

(2) I use the pack 4.5 component (. 16) to launch the 8330 emulator.

(3) I have a phone of Verizon 8330 with CDMA, V4.5.077 (Platform 3.2.0.51)

(4) I'm signing certs installed properly for RBB, RRT and CPR.

(5) I read the examples for headphones to phone, phone call and the Injection of the event.

(6) I tried to manually set permissions in options > advanced

I'm trying to write an app that interrupts a call initiated by the user and redirects to another number.

This works very well in the emulator.

However, the following code snippet generates an error of "No sig of 0 x 33" on the 8330. This occurs even if the application is duly signed.

EventInjector.KeyCodeEvent releaseEndKey=new EventInjector.KeyCodeEvent(KeyCodeEvent.KEY_UP,(char)Keypad.KEY_END,KeypadListener.STATUS_NOT_FROM_KEYPAD,100);

EventInjector.invokeEvent(pressEndKey);

But curiously

System.out.println("the injector perm is " + ApplicationPermissionsManager.getInstance().getApplicationPermissions().getPermission(ApplicationPermissions.PERMISSION_EVENT_INJECTOR));

indicates the value of 999 perm which apparently means it's allowed.

So my only guess is that Verizon phones do not allow outbound calls to be interrupted by program.

Can someone confirm or deny or otherwise tell me please what the hell I am doing wrong?

CODE COMPLETE BELOW IN THE CASE WHERE IT IS USEFUL

SDdemo.java

import net.rim.blackberry.api.invoke.Invoke;
import net.rim.blackberry.api.invoke.PhoneArguments;
import net.rim.blackberry.api.phone.*;
import net.rim.device.api.applicationcontrol.ApplicationPermissions;
import net.rim.device.api.applicationcontrol.ApplicationPermissionsManager;
import net.rim.device.api.system.EventInjector;
import net.rim.device.api.system.KeypadListener;
import net.rim.device.api.system.EventInjector.KeyCodeEvent;
import net.rim.device.api.ui.Keypad;

import java.util.Timer;
import java.util.TimerTask;

public final class SDdemo extends AbstractPhoneListener {

  private String mTarget;
  private Timer mTimer;
  private CallTask mCallTask;

  private static SDdemo mInstance;

  static public void main(String[] args)
  {
    SDdemo.registerOnStartup();
  }

  static private void registerOnStartup()
  {

    System.out.println("@@@@ phone listener start up");
    mInstance= new SDdemo();
    Phone.addPhoneListener(mInstance);

    requestApplicationPermissions();

  }
  private SDdemo(){}

  private void checkCall(String ehandler, int callid)
  {
    PhoneCall callInfo = Phone.getCall(callid);

    if ( callInfo != null ) {
      /*
       * Event Handler: ehandler
       * Telephone No.: callInfo.getDisplayPhoneNumber();
       * Elapsed Time : callInfo.getElapsedTime();
       * Call Status  : callInfo.getStatusString();
      */

        System.out.println("@#@#@#  the call is " + callInfo.getDisplayPhoneNumber());
        System.out.println("2@#@#@#  the status is " +  callInfo.getStatusString());
        System.out.println("3@#@#@#  the handle is " +  ehandler);

        mTarget = callInfo.getDisplayPhoneNumber();

    }
  }

  static private void requestApplicationPermissions() {
          //      Set permissions for the application

      System.out.println("@@@@@@ We are now requesting premissions");

      try{
          ApplicationPermissions ap1 = ApplicationPermissionsManager.getInstance().getApplicationPermissions();

          System.out.println("the injector perm is " + ap1.getPermission(ApplicationPermissions.PERMISSION_EVENT_INJECTOR));

          ApplicationPermissions ap2 = new ApplicationPermissions();
          boolean needPermission=false;
          if(ap1.getPermission(ApplicationPermissions.PERMISSION_PHONE)!=ApplicationPermissions.VALUE_ALLOW ){
              needPermission=true;
              ap2.addPermission(ApplicationPermissions.PERMISSION_PHONE);
          }
          if(ap1.getPermission(ApplicationPermissions.PERMISSION_EVENT_INJECTOR)!=ApplicationPermissions.VALUE_ALLOW ){
              needPermission=true;
              ap2.addPermission(ApplicationPermissions.PERMISSION_EVENT_INJECTOR);
              ap2.addPermission(ApplicationPermissions.PERMISSION_IDLE_TIMER);
              ap2.addPermission(ApplicationPermissions.PERMISSION_CHANGE_DEVICE_SETTINGS);
          }
          if(needPermission){
             // Dialog.inform("Please save the permissions manually");
              boolean permission=ApplicationPermissionsManager.getInstance().invokePermissionsRequest(ap2);
            //  if(!permission){
            //                                  Status.show("Application is exiting",3000);
            //      myApp.exitApplication();
            //  }
          }
        }catch(Exception e){
              System.out.println("~~~Exception while setting permissions"+e);
        }
    }

  // A call has been added to a conference call
  public void callAdded(int callId)
  { checkCall("callAdded", callId); }

  // User answered a call
  public void callAnswered(int callId)
  { checkCall("callAnswered", callId); }

  // Conference call established
  public void callConferenceCallEstablished(int callId)
  { checkCall("callConferenceCallEstablished", callId); }

  // Network indicates a connected event
  public void callConnected(int callId)
  { checkCall("callConnected", callId); 

    System.out.println("Sending DTMF for " + mTarget);

    Phone.getActiveCall().sendDTMFTones(mTarget);

    System.out.println("DTMF SENT");
  }

  // Direct-connect call connected
  public void callDirectConnectConnected(int callId)
  { checkCall("callDirectConnectConnected", callId); }

  // Direct-connect call disconnected
  public void callDirectConnectDisconnected(int callId)
  { checkCall("callDirectConnectDisconnected", callId); }

  // Call disconnected
  public void callDisconnected(int callId)
  { checkCall("callDisconnected", callId); }

  // User ended call
  public void callEndedByUser(int callId)
  { checkCall("callEndedByUser", callId); }

  // Call has been placed on "hold"
  public void callHeld(int callId)
  { checkCall("callHeld", callId); }

  // New call has arrived
  public void callIncoming(int callId)
  { checkCall("callIncoming", callId); }

  // Outbound call initiated by the handheld
  public void callInitiated(int callid)
  { 

      System.out.println("@@@@ Call init");
      System.out.println("the injector perm is " + ApplicationPermissionsManager.getInstance().getApplicationPermissions().getPermission(ApplicationPermissions.PERMISSION_EVENT_INJECTOR));

      System.out.println(mTarget.length() + " len PBX" + mPBXcall);

      if (mTarget.length()==4)
      {

          EventInjector.KeyCodeEvent pressEndKey=new EventInjector.KeyCodeEvent(KeyCodeEvent.KEY_DOWN,( char)Keypad.KEY_END,KeypadListener.STATUS_NOT_FROM_KEYPAD,100);
          EventInjector.KeyCodeEvent releaseEndKey=new EventInjector.KeyCodeEvent(KeyCodeEvent.KEY_UP,(char)Keypad.KEY_END,KeypadListener.STATUS_NOT_FROM_KEYPAD,100);

          System.out.println("###@@  calling inject 1");

         EventInjector.invokeEvent(pressEndKey);
          System.out.println(" ###@@ calling inject 2");

         EventInjector.invokeEvent(releaseEndKey);
          System.out.println("###@@ calling inject 3");

         EventInjector.invokeEvent(pressEndKey);
          System.out.println("###@@ calling inject 4");

         EventInjector.invokeEvent(releaseEndKey);
          System.out.println("###@@ calling inject DONE");

          mTimer = new Timer();
          mCallTask = new CallTask(mTarget);
          mTimer.schedule(mCallTask, 2000);

      }       

  }

  // Call removed from a conference call
  public void callRemoved(int callId)
  { checkCall("callRemoved", callId); }

  // Call taken off of "hold"
  public void callResumed(int callId)
  { checkCall("callResumed", callId); }

  // Call is waiting
  public void callWaiting(int callid)
  { checkCall("callWaiting", callid); }

  // Conference call has been terminated
  // (all members disconnected)
  public void conferenceCallDisconnected(int callId)
  { checkCall("conferenceCallDisconnected", callId); }

  // Call failed
  public void callFailed(int callId, int reason)
  {
    checkCall("callFailed", callId);

    // determine reason
    switch( reason ) {
      case PhoneListener.CALL_ERROR_AUTHORIZATION_FAILURE: break;
      case PhoneListener.CALL_ERROR_CALL_REPLACED_BY_STK: break;
      case PhoneListener.CALL_ERROR_CONGESTION: break;
      case PhoneListener.CALL_ERROR_CONNECTION_DENIED_BY_NETWORK: break;
      case PhoneListener.CALL_ERROR_DUE_TO_FADING: break;
      case PhoneListener.CALL_ERROR_EMERGENCY_CALLS_ONLY: break;
      case PhoneListener.CALL_ERROR_FDN_MISMATCH: break;
      case PhoneListener.CALL_ERROR_GENERAL: break;
      case PhoneListener.CALL_ERROR_HOLD_ERROR: break;
      case PhoneListener.CALL_ERROR_INCOMING_CALL_BARRED: break;
      case PhoneListener.CALL_ERROR_LOST_DUE_TO_FADING: break;
      case PhoneListener.CALL_ERROR_MAINTENANCE_REQUIRED: break;
      case PhoneListener.CALL_ERROR_NUMBER_NOT_IN_SERVICE: break;
      case PhoneListener.CALL_ERROR_NUMBER_UNOBTAINABLE: break;
      case PhoneListener.CALL_ERROR_OUTGOING_CALLS_BARRED: break;
      case PhoneListener.CALL_ERROR_PLEASE_TRY_LATER: break;
      case PhoneListener.CALL_ERROR_RADIO_PATH_UNAVAILABLE: break;
      case PhoneListener.CALL_ERROR_SERVICE_CONFLICT: break;
      case PhoneListener.CALL_ERROR_SERVICE_NOT_AVAILABLE: break;
      case PhoneListener.CALL_ERROR_SUBSCRIBER_BUSY: break;
      case PhoneListener.CALL_ERROR_SYSTEM_BUSY_TRY_LATER: break;
      case PhoneListener.CALL_ERROR_TRY_AGAIN: break;
      case PhoneListener.CALL_ERROR_USER_BUSY_IN_DATA: break;
      case PhoneListener.CALL_ERROR_USER_BUSY_IN_PRIVATE: break;
      case PhoneListener.CALL_ERROR_USER_NOT_AUTHORIZED: break;
      case PhoneListener.CALL_ERROR_USER_NOT_AVAILABLE: break;
      case PhoneListener.CALL_ERROR_USER_NOT_REACHABLE: break;
    }
  }
}

CallTask.java

import java.util.TimerTask;

import net.rim.blackberry.api.invoke.Invoke;
import net.rim.blackberry.api.invoke.PhoneArguments;

public class CallTask extends TimerTask {

    public String mCallTarget;

    public CallTask(String _target)
    {
        mCallTarget = _target;
    }

    public void run() {

        System.out.println("calling task dial");

        PhoneArguments call = new PhoneArguments(PhoneArguments.ARG_CALL,"555-867-5309");
        Invoke.invokeApplication(Invoke.APP_TYPE_PHONE, call);

    }

}

Thank you for pointing me in the right direction almeida.

In case this help someone else:

(1) looking at conducting thread me section of the knowledge base on problems with "52525400 = RIM Runtime API" line missing in the CSL file when using JDE.

(2) I use Eclipse JDE plugin. When I looked in the file AREA, this tag was present.

(3) poking around a little more, I noticed this tag was missing from the CSO file. So I added, why not?

(4) things like magically started working. I go out, it stops working again.

For Eclipse IDE using JDE plugin, I'm so going on a branch and saying: you must add '52525400 = RIM Runtime API' to your CSO file.

to make a call in a different way

Normally when you place a call I did like this:

Pa = PhoneArguments

new PhoneArguments (PhoneArguments.ARG_CALL, newNumber) ;

Invoke.invokeApplication (Invoke.APP_TYPE_PHONE, pa);

What I wonder if it is possible to call the phone first and then send dtmf rather tones?

Yes, you can call the application phone to phone and then inject the call of DTMF tones. After invoking the phone application to dial a number using the Phone.getActiveCall () method to get an instance of phone call, then use it to call the methods sendDTMFTone (s).

Skype sends more of his keyboard DTMF tones

Yesterday, I noticed that I could access is no longer my daily teleconference as the remote system did not accept the access code I'm punching in Skype keyboard.

Today, I tried to access a menu system (coincidentally, for the MS customer service). I reinstalled the latest version, but it still does not work.

The problem occurred from the Click to Call plugin, breaking the last revision of browser Chrome uninstall.

Is there a known here fault or finally has something broken on my machine?

(Windows 8, office)

Thanks for this - by acting on this information, I decided to try to call my own phone and hear the tones, since I'm sure that no dial tone was sent, and not a question of quality (that I don't have from time to time).

In doing so, I found my camera is not plugged in, so no micro. This must have disabled all sounds, including Skype by sending sound signals.

Connected to the camera and I'm back in business.

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.

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.

Send the number to another device

Dear all,

I have a question by sending the number to another device. In my application, I use the DTMF to send the antother device number. But the speed is slow, is there not another method to send the number, such as fsk?

Thank you for your help

Leo

I would say that you use a digital format to transfer data, such as http, mail, etc.

He has only dtmf on blackberry.

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.

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.

Send dtmf or wherelse

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