dtmf ivr

practical information. The text message service provides guidance for busy people to obtain the most effective information in the most time-saving way, and for those who want more detailed information, it can be implemented through WAP and MMS, which, to a certain extent, drives the use of the first two forms. 4. IVR: the best way to disseminate mobile phone information for users The "mobile phone newspaper" in the form of

implemented in the timer interrupt. The timing parameter should be selected taking into account the various timing requirements in various operations of the system. The timing frequency is generally an integer multiple of the minimum timing frequency. Any software latency is not allowed. The key to implementing time-sharing control is to reasonably break down the functions of the system into various job modules. The finer the job module, the better the real-time performance of the system. Altho

is called to obtain a socket Number: Int socket =: socket (ss. ss_family, SOCK_DGRAM, 0 ); This socket number will be assigned to the mNative variable of the Java layer: Env-> SetIntField (thiz, gNative, socket ); The port number is directly returned by the create FUNCTION. The create FUNCTION supports IPv6. Voice Stream: AudioStream Android.net. rtp. AudioStream is inherited from RtpStream and represents a voice stream that is built on the RTP protocol to communicate with the other party. A

the module.There are two kinds of melody available: Arrival voice, data or fax call melody and arrival text message sound.19. At + crsl sets or gets the sound level of incoming phone ringtones. Ii. Call Control commands1. ATD dialing command. This command is used to set a call, data, or fax call.2. the ath host command.3. ata answers the phone.4. At + ceer extended Error Report. This command is used to explain why a call is interrupted when the call setting fails.5. At + vtd provides users wit

available: voice callsand data or fax calls to melody and incoming SMS sounds.19. AT+CRSL Sets or gets the sound level of incoming phone tones.Two. Call control commands1. ATD dialing Commands. This command is used to set up calls. Data or fax calls.2. ATH Hang command.3. ATA answer phone.4. At+ceer Extended Error Reporting. This command gives the reason for interrupting the call after the last call setup failed.5. AT+VTD provides users with the application of GSM network to send

. This command plays a piece of music on the buzzer of the module. There are two kinds of melody available: Arrival voice, data or fax call melody and arrival text message sound. 19. At + crsl sets or gets the sound level of incoming phone ringtones. Ii. Call Control commands1. ATD dialing command. This command is used to set a call, data, or fax call.2. the ath host command.3. ata answers the phone.4. At + ceer extended Error Report. This command is used to explain why a c

initiated in the same conversation after the initial invitation are considered re-invited (Reinvite). Configuring Canreinvite=no allows the asterisk media channel to pass through itself without allowing RTP information to be transmitted directly between endpoints. Asterisk does not initiate a re-invitation under any of the following circumstances: If either side of the client is configured as Canreinvite=no, Asterisk needs to perform a speech encoding conversion if the client cannot negotiate t

255.255.255.0 10.22.110.254 Ip route 10.22.109.0 255.255.255.0 10.22.110.254 Ip http server Ip pim bidir-enable ! ! Call rsvp-sync ! Voice-port 1/0: 0 ! Voice-port 1/0: 1 ! ! Mgcp profile default ! Dial-peer cor custom ! ! ! Dial-peer voice 10 voip Description Phone to Fenzhi1 Destination-pattern 6964 .... Session target ipv4: 10.22.118.10 Dtmf-relay cisco-rtp h245-signal h245-alphanumeric Fax rate 9600 ! Dial-peer voice 20

: Select, read or test service group (Fax:select, RE AD OR TEST SERVICE CLASS) At+fmi Telex: report produced identification (Fax:report manufactured ID) AT+FMM Telex: Reporting mode ID (Fax:report model ID) AT+FMR Telex:The report amends the identification (Fax:report REVISION ID) at+vtd (TONE DURATION) At+vts DTMF and the generation of the dial tone (DTMF and TONE GENERATION) At+cmux multi-channel control

', ToString (k), ToString (v))) End for K, V in pairs (obj) does printsessionfunctions (obj) print (String.Format (' obj k->%s v->%s\n ', ToString (k ), ToString (v))) End If _type = = ' table ' then for K, V-pairs (_type) do print (String.Format (' _type k-gt ; %s v->%s\n ', ToString (k), ToString (v)) End end print (String.Format (' \ n (%s = = DTMF) and (obj.digit [%s] = = [%s] Tablag RABAMOD.DTMF) \ n ', _type, Obj.digit, TABLAGRABAMOD.DTMF) End

means that the IRQ position is represented in ISR, and the corresponding position in IRR is zero, indicating that the interrupt is being processed by the CPU. At the same time, write its number to the lower three digits of the interrupt vector register IVR (IVR is specified by icw2. Do you still remember that the lowest three digits of icw2 are 0 at the specified time, here, they are used !) At this time,

implementation Acd. Agent Login and routing services, providing control level, interface level two development package AP. Agent Access Proxy for the ACD to share the pressure Ims. Unified Call Model Maintenance Ivr. Self-service voice interactive platform, providing visual customization of the IDE RECORD. On-demand/automatic distributed call recording Logical architectureTypical networking scenariosManual Feature List

= true' in braces {}. For example: {local_var_clobber = true, sip_secure_media = true} Sofia/default/[email protected] | Sofia/default/[email protected] | [sip_secure_media = false] Sofia/default/[email protected] 9. Configure FS to call external GatewayCreate gw1.xml content under CONF/sip_profiles/external 10. Implement the IVR language menu www.freeswitch.org.cn/2010/03/21/yong-freeswitchshi-xian-ivr.htmlFirst, modify CONF/dialplan/public. xml if

. */void Dial (String number); /** * Place a call to the specified number. * @param number The number is called. */void Call (String number); /** * If There is currently-a call-in-progress, show the call screen. * The DTMF Dialpad may or May is visible initially, depending on * whether it is up when the user last exited th E Incallscreen. * @return True if the call screen is shown. * * Boolean showcallscreen

promote the depth of the content of the Bell Sound development, rich with characteristics of the music box monthly products; (5) Appropriate user needs, the user can perceive the quality of service as the starting point, and continuously refine and improve the IVR guide, seven tones of tone website, SMS push function. Channel current competitor provides the color ring service processing channel to cover the text message,

Recently the call center went mad, my "step by Step Development Call Center" series in the process of writing, encountered a variety of problems, tonight, to record a tangle of my n-long problem: When the inner line calls out through the card, if the other call center needs to send the key response (such as the need to enter the phone number when dialing 10086), call the API function SSMTXDTMF regardless of the other side prompt error. Find n multiple solutions, contact n multiple

Address: http://www.pjsip.org/pjsua.htm Introduction Pjsua is an open-source command line SIP User proxy (soft phone), which is implemented using pjsip protocol, pjnath, and pjmedia. Although it only has a simple command line interface, it is fully functional. SIP function: Multiple IDs (account registration );Multiple calls;Support IPv6 (added in version 1.2 );Prack (100rel, RFC 3262 );Update (RFC 3311 );Options;Call persistence;Call transfer;Simple pidf and IDF support (subscription/no

gets the sound level of incoming phone ringtones. NoneIi. Call Control commandsATD dialing command. This command is used to set a call, data, or fax call.Ath host command.ATA receives the call.At + ceer extended Error Report. This command is used to explain why a call is interrupted when the call setting fails.At + vtd provides users with DTMF (dual-Tone Multi-frequency) Dual-audio sent using the GSM network. This command is used to define the length

screen. Similarly, if you transmit the DTMF call number information, you only need to use the DTMF decoder chip to decode the data to display the DTMF call number. ...... Omit irrelevant content ...... The above is a very small part of the telephone communication protocol. The materials are too long. Do you think it is a bit dizzy? It doesn't matter. I will giv

are executed in a single cycle. The throughput can meet the requirements of various new network connected applications, and the program flash memory can also provide online and offline re-programming, as shown in figure 1 of the specific structure of IP2022. ② The role of DSP in network speech products is irreplaceable. It mainly completes audio and video coding and decoding. Therefore, when selecting a DSP, we must consider meeting the current basic requirements, such as session functions and