Php code to get calleridроботи

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    309 php code to get callerid робіт знайдено, ціни вказані в USD
    Asterisk Call API 6 дні(-в) left
    ПІДТВЕРДЖЕНО

    Hello, I need an application like this: http://ip/[увійдіть, щоб побачити URL]{CALLERID}&call={CALL} If I request this web address, Asterisk will: 1-) Call {CALL} via {CALLERID}. 2-) After CALL answers, send call status (busy, rejected, answered etc), duration and recorded soundfile path to our REST API service. - Need support over 50 calls at same

    $173 (Avg Bid)
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    Project for Sudesh A. Закінчився left

    I know this, but the app is almost complete. I just need to change the JSON response to id instead of idCall using this script [увійдіть, щоб побачити URL] Screen Shot 2018-10-31 at [увійдіть, щоб побачити URL] Shot 2018-10-31 at [увійдіть, щоб побачити URL] Screen Shot 2018-10-31 at [увійдіть, щоб побачити URL] Shot 2018-10-31 at [увійдіть, щоб побачити URL]

    $120 (Avg Bid)
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    ozeki callerid Закінчився left

    i need to set calledid for outgoing call that all i need

    $31 (Avg Bid)
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    adroid callerid to web service Закінчився left

    I need a adroid app to read caller id of a cell pohne and send that id to a web servce or a database with wifi in local netwok

    $190 (Avg Bid)
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    We have an on-prem, small PHP webapp running on IIS (Server 2016) that looks up a telephone number and returns the result to our phone system. Our phone system routes incoming calls based on whether the CallerID number exists in a database, what type of company (client, vendor, unknown). When we migrated the application to Windows Server 2016, some

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    We have A2billing server and we are building an app for our resellers to topup customers through app. We need the working API to Create and Manage customers - mainly Create customer with Callerid (including sip account) and topup.

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    I have a working FreePBX server. The freepbx server is running good so far. The following changes needed in my server. 1. I want to install a open source predictive Dialer in my freepbx. I have have chosen VICIDIAL for that. You may suggest better one. The dialer must use existing extensions for auto dialing features. 2. You must configure the system

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    ...gateway. The GSM gateway is working good so far. But I need a small changes in configuration for smooth operation of the gateway. The gateway is not passing correct caller id to my PBX. Caller ID has two parts. One is Caller ID name and another is Caller ID number. Caller ID name part is okay and gateway sends is as the actual caller ID. Where is Caller

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    ...500 Field Salesmen on their mobile phones. I am looking to get an app developed for recording as well as taking dictation notes of incoming calls. Requirement in a nutshell is: 1. App in Android only (No IOS) 2. App wakes up on incoming call (like call recording apps). 3. App listens to the conversation and does 4 things a) Transcribe what the caller

    $528 (Avg Bid)
    Угода про нерозголошення
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    asterisk cdr and cel parsing Закінчився left

    we need to get: missed calls with callerid/number completed calls with callerid/number and recording (recording already enabled) from cdr and cel database. our setup asterisk FreePBX [увійдіть, щоб побачити URL] Call Logic: timegroups-timeconditons-queues

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    ...JSON/NODEJS/PAMI/PAGI/ARI etc... One Way Functions : (Web service will send data) - Send CallerID (Customer will set an URL from pbx panel. Web service will send about call status(Extension ringing/answered) for per extension)(Calls may be get from queue or IVR or another, i need to alls) - Online Calls (Customer will set an URL from pbx panel. Web service will

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    Adding the I frames for asterisk server to send real time called information to an API. We need: Caller ID, Callers name, Call Statuss, etc.... Example: CALLERID=2125555555&CALLERNAME=&last_name=&EXTINFO=XpressTrainner&AGENTINFO=XpressTrainner&list_id=999&phone_code=1&title=&first_name=&middle_initial=&last_name=&address1=&...

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    3CX Flow designer - custom app Закінчився left

    Hello We want an app of the 3CX Call Control API in the 3CX server, listen to the events related to calls, and when a call ends, trigger the notification (post callerid&Queue Name to our webservice).

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    Терміновий
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    Code must be compatible with PHP 7.x and WHMCS v7.4.2 $ v7.5 Create a program which is displayed within the Client Area of WHMCS that allows a client who has an active product server of VoIP (which already exists) to enter CallerID Numbers to be filtered and added to a new table {mod_CallerIDFiltering}. Will use existing APIs to send commands to

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    Asterisk - CTI (Click To Call) API This will work on a Asterisk 13, FreePBX 13, and Asternic Pro 2.1.x platform. We need to provide a REST API for several actions e.g. a) Click To Call - e.g. /[увійдіть, щоб побачити URL] authid = preset hash value to authenticate the user against. Return in JSON - Asterisk

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    Asterisk Dial Plan Rule Закінчився left

    ...Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1) We need something like the following code added to the general rule, so that from the SIP Header the CID "+" is stripped off and changed to "00". [from-trunk] exten => _X!,1,GotoIf($["${CALLERID(num):0:2}" != "+1"]?noplusatstart) exten => _X!,n,Set(

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    Build me a SIP program Закінчився left

    SIP Program (Really simple) Create, edit, delete users & each user has one-more parameter named URL. when call inbound to user the program knows send GET request with callerid to specific assigned user's URL. GUI is not required, you can make it with NodeJS or C# or whatever. it should be SIP server for anything. (the program)

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    ...API ( I will provide login credentials of an Twilio account for testing purposes ) which will place multiple phone calls every X seconds ( where X is a Customizable VARIABLE ) to a PRE-DEFINED list of phone numbers and once the call is being picked up the script you play a customizable voice/music file ! Therefore the application must have the following

    PHP
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    ...commands. 1) One command is used to send fax out via SIP trunk directly without using IP-PBX like Asterisk ...etc. I need one command similar to the previous project mentioned in the following URL link. https://www.freelancer.cn/projects/c-programming-linux/utility-send-fax/ 2) The 2nd command is used to receive fax via SIP trunk directly

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    Customize Issabel report Закінчився left

    ...transfer the call to an evaluation system "specific number" the customer will hear an IVR with the ability to choose "ex. How satisfied are you with the customer service staff, choose 5 if you are bery satisfied and 1 if sure not satisfied" another IVRs with questions and answers, and in the end there is an option to record a voice note "ex. to...

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    make auto callerID thing Закінчився left

    need new feature in dialing app to auto pick caller ID before making a call.

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    ...number and would like the TwiML Bin to do the following: Simulring multiple phones (start with two, when a customers calls the 1800 number, it calls multiple phones, whoever picks up first gets the call) timeout 60 seconds (call goes to mp3 recording "RecordingUrl" after 60 seconds of no answer) callerId (so the user being called can see 1800

    PHP
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    Twilio API TWIML Voice TwiML Bin Закінчився left

    [увійдіть, щоб побачити URL] Twilio TWIML Voice TwiML Bin Simulring multiple phones timeout 60 seconds timeLimit 60 mins callerId record-from-answer-dual RecordingUrl (play mp3 after 60 seconds of no answer)

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    Twilio TWIML Voice TwiML Bin Закінчився left

    [увійдіть, щоб побачити URL] Twilio TWIML Voice TwiML Bin Simulring multiple phones timeout 60 seconds timeLimit 60 mins callerId record-from-answer-dual RecordingUrl (play mp3 after 60 seconds of no answer)

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    Asterisk AMI GUI Закінчився left

    ...on our plant; By connecting via AMI we want to manage the events from the interface. We want to post event url via GUI. Example process: Extension ringing Extension Call Answering Extension Call Termination We want to POST values like CallerID Call Unique in these events. We want to manage these operations from the relevant interface

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    FusionPBX API assistance Закінчився left

    ...basic requirements to assist us to integrate into our CRM. We do not need an integration developed, we simply need examples of the best practices to make the API calls we need. Firstly a prerequisite is that all calls must be recorded. Extensions are set to record all. 1a. API call to originate an call from an internal extension to an external pho...

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    Dear Sirs i would like when someone calls, the sip phones to show up his name that i will have previously stored in the freepbx phonebook page i tried in the "Caller ID Superfecta" page of freepbx to enable the "Asterisk Phonebook" and in "CallerID Lookup Sources" to select source type:"internal" but nothing changed Your ...

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    home pbx Закінчився left

    Home project. I have given up on using these voip desktop phones without a PBX. I need someone to configure "Incredible PBX" (atop Ubuntu or CentOS) for 4 Cisco SPA525G2 phones, and 1 Yealink SIP-T48G. The PBX will live in the garage with pfSense. I do not use wifi for any device other than smart phones and tablets. I will be keeping my existing

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    C# SIP Protocol CallerID Client Закінчився left

    I'm looking for a basic SIP client implementation written entirely in C# from scratch. The client does not need to be capable of actually establishing voice or video calls, I actually just need it to be able to register as a SIP client to the provided server (using username/password authentication) and then pop a notification on the screen when there

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    PHP & WebRTC Coding Закінчився left

    We have first project , we need existing o...have first project , we need existing outbound web app to auto select from existing list of purchased callerID #'s, based upon the list selected to call. We require all work to be done with remote view tool such as AnyDesk.com. Must be reliable, with fast internet please. Must be EXPERT in PHP/WebRTC.

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    ...dialer needs auto selector for callerID to match outbound contact list area code's. We already have a drop down menu that shows the available/purchased callerID's so all the info is present. Expert must have fast internet, stable electric, and TeamViewer to do work only. We do not provide unsupervised server access to do work. All work must be done

    PHP
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    ...for call forward but it's not adding the correct data into the CDR dst. I want (CDR(dst)) to be the number the call was forward to. I don't want another column added to the CDR table, has to be the current column. [subInbound] exten => s,1,Set(thedid=${CALLERID(dnid)}) exten => s,2,Set(theext=${ARG1}) exten => s,n,GotoIf(${DB_EXISTS(C...

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    Прихований
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    ...find below project requirement which we need to add in the fusion pbx 4.0 running on free switch.1. Project Scope:Enable API setting to send inbound caller ID calling party information for any domain/tenant as fully supported solution.2. Use case:Client want to send a caller id information through API to Icabbi for domain configured inside fusion pbx

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    ...find below project requirement which we need to add in the fusion pbx 4.0 running on free switch.1. Project Scope:Enable API setting to send inbound caller ID calling party information for any domain/tenant as fully supported solution.2. Use case:Client want to send a caller id information through API to Icabbi for domain configured inside fusion pbx

    $267 (Avg Bid)
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    Hi, I need a sample code to identify the callerID, SMS phone number for an incoming call. I believe with iOS 10, this is possible so I am looking forward to have a sample code for it. My budget is 100USD fixed.

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    Setup Asterisk PBX and chan_dongle Закінчився left

    We use chan dongle on an rabpberry board with 3 x gsm huawei we setup all for working with sms ---- we want to setup for calling and receive call from this 3 lines setup robot voice and callerid we use eyebeam to connect to voip

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    BUILD AN IVR APPLICATION ON 3CX PBX Закінчився left

    We want an application to run on 3CX PBX This application will automate the following process 1) A customer calls in the PBX 2) The PBX answers with a welcome message and puts the customer on hold 3) The PBX looks up a webservice by passing the customer "callerid" to retrieve a "destination phone" 4) If the destination phone is not recovered

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    Будь ласка, зареєструйтесь або увійдіть в систему для перегляду деталей.

    Обраний Прихований

    FreePBX 13 PHP 5.6 Asterisk 13 Repair of small php script and code a new WEB portal to display results from a MySQL table. I have a small script that collects key input for a post call survey, I can’t get the agi_channel added to the MySQL table. This needs to be fixed. $agi->request["agi_uniqueid"]."', "....

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    Прихований
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    Asterisk Post Call Survey Закінчився left

    The call will being transfered from any extension to this IVR. I need to build a post call survey to make customers 3 questions and save response (dtmf) into mysql. Audios will be prerecorded audios. Also need to associate to the same data: callerid, time, date.

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    Asterisk Call Results Script Закінчився left

    Brief: Asterisk Call Results Asterisk 13 PHP 5.4 Simple agi script using PHP to output call results to two MySQL tables, each table on different MySQL servers, one table showing the Status by tailing the “messages” log file, looking for the string “#########@DID” of original number “Set: number=###########” and the other MySQL T...

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    Прихований
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    i want to setup a gsm voip gateway GUI using asterisk+chan dongle, use huawei e173u-1 and e153u-1 e169. i already have a running server ( to learn from), but i want to setup a new server with the following features in a GUI. those features are reading output from CLI commands and text from responses of sent ussd codes . i just need to represent

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    a2billing fix Закінчився left

    ...registered on a freeswitch and when the client makes a call, the call goes to the a2billing to verify credit. I'm trying to setup a second server and imported the database from the original one. All customers are there, all callerid, all rates, everything looks right but all calls to that server fail and when I generate the [увійдіть, щоб побачити URL]

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    ...those to a localhost http server by calling a link every time the call state changes. <OFF> I already wrote a similar program once 6 month ago under Delphi XE8 ... but it does not run now on Android 6.0 (and never runned under windows). So it would be logical to rewrite under Xamarin because MS claims continuous free upgrades... I have tried to write

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    Обраний
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    Project for gilbertarias Закінчився left

    Hi gilbertarias, I have an issue with Elastix: When forwarding calls, the original callerid is replaced with the main office DID. Thanks, Ruben

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    Build an Asterisk Dial Plan Закінчився left

    ...trunk) CallerID for A will be fetched and set from MYSQL table randomly. When B answers the call we start a conf bridge and put A and B to this conf bridge (quiet mode) and include a third silent party which will record B Leg only After a certain time (2-3 mins) we will forcely hangup B Leg and stop the recording And play a wav file to A side in

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    Napisz oprogramowanie Закінчився left

    Witam, Poszukuje osoby umiejącej zarządzać serwerem PBX oraz znającą protokół SIP. Potrzebuje stworzyć kilka linii z konkretnym CallerID (Numer bądź nazwa) Coś jak spoofing numeru.

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    $26 - $219
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    Asterisk record Queue Calls Закінчився left

    ...start recording when customer is bridged (talking) to agent. We also run “Queue Callbacks” so Exten and CID might be mixed. MixMonitor(/var/spool/asterisk/monitor/${YEAR}/${MONTH}/${DAY}/Outbound-Agent-${AGENT}-${CALLERID(num)}-${UNIQUEID}.WAV) Only record calls that originate from a Queue. To include on string Exten(Agent) Queue number, CID

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    SETUP OPENSIPS ON AMAZON EC2 Закінчився left

    ...OPENSIPS on Amazon EC2 server to authenticate sip users with username and password and receive all the calls and route them to a2billing server. I have a2billing server set-up but it will need configuring for opensip sip peers to connect and authenticare using CallerID. So SIP peers with dynamic IP will send traffic to Opensips. Opensips will authenticate

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