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    13,427 sip configuration symbian tugasan ditemui, harga dalam USD

    ...APPBuilder that allows to include the SIP / VOIP SDK to connect to ASTERISK as an option and if it were not possible in ANDROID Studio The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Call from a number of clients (via Json) 4.- Configure (add / modify / delete) SIP accounts 5.- Management Module

    $30 - $250
    $30 - $250
    0 bida
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    ... Its all about Fun Art, not Fine Art! And remember, we are a BYO studio, so bring along your favourite bottle or two of liquid creativity to enjoy while you paint and sip with us on the day. You supply the wine and we’ll provide all the art materials, glassware and good vibes. Leave the rest to us! This is not your typical painting class

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    12 penyertaan

    looking for an artist/painter/instructor to teach kids and art classes/parties for different ages. 3-5 hours a day, 2 times a week. Ideal candidate should be positive, patient, energetic. Looking to hire immediately.

    $25 (Avg Bid)
    Tempatan
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    ...gin=gs102&phone_pass=865748&fronter=321&closer=321&group=MORTCP&channel_group=MORTCP&SQLdate=2018-05-22+113857&epoch=1527014338&uniqueid=1527014305.751&customer_zap_channel=SIP/enterprise-000000ff&customer_server_ip=&server_ip=69.64.71.182&SIPexten=gs102&session_id=8600056&phone=7029973001&parked_by=55502&dispo=&dialed_n...

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    7 bida

    ...not the best with Asterisk and I am unsure how the control panel would communicate with Asterisk. This is what we need the control to do: - Add a SIP Trunk Channel (For PSTN connectivity) - Remove a SIP Trunk Channel - Add Telephone Numbers to the database which can be used by extensions for outbound and inbound calls - Remove telephone numbers - Create

    $733 (Avg Bid)
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    ...can be achieved with a call conference or call bridging. when dialing user (A) calls user (B) they are put into a call conference or bridge. if the dialer(user a) hangs up(sip 487) before user b answers, a disconnects but the conference / bridge channel stays open until b hangs up or for a max of 30 seconds cli needs to be passed from a to be etc

    $196 (Avg Bid)
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    3 bida

    We need sip protocol writer. There is sip application for testing for voice also need recording of voice on app

    $540 (Avg Bid)
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    18 bida

    I need a python script to: 1. answer a SIP call using pjsip 2. listen & send the audio in chunks to google speech 3. get the recognized text back I would like to either detect the silence to send the audio to speech service or continuous send and get the text as it is ready

    $528 (Avg Bid)
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    I am seeking a techie, who c...traffic is passed through blocked gateways. 4. Configure auto provisioning for IP Phones and mobile clients. 5. Mobile clients must be compatible across various OS. 6. Configure SIP trunk for International calls (preferably google voice). 7. Any other work that shall be required for smooth implementation and functioning.

    $419 (Avg Bid)
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    3 bida

    We are a VoIP company selling SIP trunks and Hosted PBX services. We are looking for a new modern website for our company to showcase our products and services. The clients should be able to do some basic account management on our website where the API side comes to our SIP server. Only the basics like viewing account balance and topup account with

    $300 (Avg Bid)
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    2 penyertaan

    This is a continuation of a project previously posted. We are in the final stage of configuring VTiger as a CRM to provide a lead-gen, automation & follow-up mainly for our Affiliates. Please review the description in the attached file and provide me a bid with estimated time-to-completion, expressed as the number of days you need from start to finish. PHP skill is a "must" in addi...

    $196 (Avg Bid)
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    3 bida

    We need a Phone Sales Person (English Speaking Countries) Part time for at least 20h/week. PEQRFECT ENGLISH IS A REQUIREMENT. You will work with our CRM / SIP Phone.

    $5 / hr (Avg Bid)
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    16 bida

    We would like a poster designed which will be hung on the wall of our Paint and Sip studio. The poster should be 300 dpi/high resolution as it will be printed out at about 45 inch wide to 90 inch high (so thats the ratio on size ) and hung on our wall. The poster shows a list of fun things that people might do at our studio while at our events. So

    $80 (Avg Bid)
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    35 penyertaan

    i want to create own website like mutual fund , sip and financial and admin login client login and sip calculator also available in my web site and nse,mcx stock market api also required

    $191 (Avg Bid)
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    ...Configuring an Open Source. U should have knowledge in given below application (Installation and configuration ) maven v3.2.x PostgreSQL v9.4 Elastic Search v2.4.x Jboss Wildfly v10.x Git 2.8.3 JDK 8 update 112 or higher This project is only for installation & configuration of the open source and after that we will start a new project for the customization.

    $164 (Avg Bid)
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    I'm looking for someone to develop an Outlook plugin for windows that you login with a VoIP SIP extension credentials and whenever a call comes in it searches the number in Outlook and if it finds a contact it opens up that contact in a new window. If it doesn't find a contact it opens a new contact window with filed in the details from the caller ID

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    I need you to develop some software for me. I would like this software to be developed for Windows usin...need you to develop some software for me. I would like this software to be developed for Windows using .NET. I would like a small desktop tray application to keep checking my sip provider for any voice mail messages left using wmi. Using vb.net.

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    11 bida

    ...the PJSIP library to implement Voice over IP (VoIP). This is an open source multimedia communication library written in C. The library uses a high-level API by combining the SIP (signaling protocol) with the multimedia framework and NAT traversal functionality. This API is suitable for all types of systems including mobile handsets. Group video chatting

    $1894 (Avg Bid)
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    need someone to finish home project PBX using freepbx 14. I completed the setup by the inbound calls fail. I have two test extensions I have one Sip trunk I have recordings for IVR i have one DID

    $61 (Avg Bid)
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    9 bida

    ...recorded. I need to be able to query CDR to check all communication between parties + open recorded communications. Incoming calls will come from PSTN, SIP trunk, and all outgoing call through sip trunk. Script may be based on freeswitch as it’s an xml based solution, faster for administration....

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    Our company Globetel Consultancy Service is based in Malaysia. We are looking for a candidate who can help to develop a SIP to viber/whatsapp gateway. The gateway should be able to convert the incoming voice calls over SIP and forward that through viber/whatsapp in order to complete the call to the receiver number. In addition to that it should be able

    $2196 (Avg Bid)
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    This is a repost! I need an application that can send/receive calls through SIP (Can use any sip stack like sipdroid) and forward it to the GSM network. The application should then forward the audio and convert from SIP to GSM and vice versa. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset

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    I have a shoregear 120/24 that I need reset to factory defaults, then added to an existing ShoreTel 7 system. I have a console cable hooked up to the Shoregear switch and will give you remote control access to the computer with the serial cable hooked up, then will give you access to the computer running the Director software/webpage. Need to get this done ASAP ShoreTel 7 Build 12.5.9303.0

    $30 - $250
    $30 - $250
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    i am working on an app that can provide unlimited calling from Canada to India( same functionality as Rebtel). I looking for someone who can provide pstn gat...functionality as Rebtel). I looking for someone who can provide pstn gateway solution in Canada with unlimited incoming. Any type of pstn gateway solution can be considered( ex. pri, sip......)

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    Configure and implement full functionality of website on AWS server. Write APP API. Set up MySQL database.

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    Configure SIP server and post live calls gor each DID in database. Also we need to create and update inbound and outbound via API’s

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    I need to configure SIP server which supports API's for Inbound and Outbound also i need live call data based on DID

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    1 bida

    ...are building a freepbx server in Canada that can receive inward and outward calls and need some advising setting up a pstn gateway. For example pri line would be better or a sip trunk or if there is any other option ? Looking for a knowledgeable person who can get this project started. p.s. we will be posting some more jobs for the same so will need

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    Hi I need to send email to 1.000.000 Clients all the client from france with (orange,free,sfr) email provider .

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    19 bida

    Initial Payment Conditions Price should not be more than $ 100. No payment untill job done to satisfaction. Kindly don't waste time if you cannot work within this budget and conditions. Web Site requirements Look at onlineguides.net. Need to complete it with free and some premium plugins , Learn Press. Install Woo plugins compatible with Learn Press. Make the Learning Content Read Only to ...

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    4 bida

    We have a Vanilla SIPWISE Server Community Edition. We are looking for someone skills that can setup our SIP Providers and do test.

    $227 (Avg Bid)
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    5 bida

    setting up a WordPress theme to my host including all plugins [login to view URL] including preventing theme detectors (like WPSNIFFER ) from detecting theme name Important note:::: malware and backlinks will be detected instantly by server and will block the host account Will be a reported in case we have any viruses, malware, or backlinks

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    ...custom Outlook addin to connect to a SIP extension and when an incoming calls comes in it should give a pop up notification with the contacts info from outlook. It should be exactly like this: [login to view URL] but it should be able to connect to any asterisk SIP server. You can even use this addin

    $567 (Avg Bid)
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    5 bida

    Freepbx general support, particularly security, external sip phone connections like csipsimple not working, obi202 not registering, vpn connection setup, etc

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    10 bida

    I need a plugin to be set up on my website and a page to be created. Looking to use ads manager pro plugin. [login to view URL] Looking to ad a ads section to only use that plugin only for that area of the website. So 2 tasks set up ads manager for some areas of the blog 2. Create page.

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    ...com/file/d/1S3WiUlALcNfYEKo3SENMmga9ktDUaNgb/view?usp=sharing Has anyone experienced anything similar? Please, let me know if you need more details of the set up. See below our nginx configuration listen 443 ssl; # managed by Certbot ssl_certificate /etc/letsencrypt/live/casista.com-0001/[login to view URL]; # managed by Certbot ssl_certificate_key

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    ...concept. HTML mockups for layout purposes will be provided, but layouts are to be configured for CSS according to a theme 1. Files have been uploaded to web server, but configuration still needed to complete the install 2. Set Up the Credit card processing functions, with our gateway and merchant accounts. 3. Configure Customer area to match the html

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    We need help to integrate flexisip with linphone and freeswitch. Flexisip is partially working with ...to integrate flexisip with linphone and freeswitch. Flexisip is partially working with push and outbound calls, we just can't figure out why flexisip database is not storing sip user/pass. Also need help with redis server integration with flexisip

    $563 (Avg Bid)
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    8 bida

    We need help to integrate flexisip with linphone and freeswitch. Flexisip is partially working with ...to integrate flexisip with linphone and freeswitch. Flexisip is partially working with push and outbound calls, we just can't figure out why flexisip database is not storing sip user/pass. Also need help with redis server integration with flexisip

    $139 (Avg Bid)
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    2 bida

    Unable to connect to remote asterisk (does /var/run/asterisk/[login to view URL] exist? also run the application - test sip and configure basic config if required

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    We have 3 wordpress b2b websites in the same web server from the Hosting Gator. Currently, they use the separate databases in their Cpanel. We want to use one sharable MySQL business listing database instead of the current 3 separated business listing databases, since the business listing database can be sharable by 3 websites. 1. It might need to have MySQL server by configuring it in the c...

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    Perjanjian Kerahsiaan
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    32 bida

    Run asterisk gives error while asterisk -rvvvTroubleshoot sip connectivityFix other minor issues

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    9 bida

    Hi Guys, I want to configure in my Google Tag Manager a few services and I'm looking for someone with an experience.

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    18 bida

    I need VoIP Softphone With Sip Unblocking Proxy. If you have any solution please contact me.. I need only android Version..

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    9 bida

    ...org/licenses/[login to view URL] Must be able to work with existing source primarily Java and XML and modify to specification. Familiarity with Signal and Video processing as well as SIP,XMPP,RTSP and streaming (RTP,codecs) Final result must be a working APK capable of being loaded directly to the phone without Google Play Store and also downloadable from Google

    $314 (Avg Bid)
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    9 bida

    I don't think it would be a big proje...a big project just a simple web portal for end-user customers to make changes to the Voice routing in phone numbers without the Twilio branding. The numbers normally go to a SIP trunk but the end-user (my customer) wants the ability to self-change the phone number destinations to redirect to a cell phone number.

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    Write on python 2 or 3 simple script: -Connect to sip server with credentials (username, password from sip) -wait for call -answer a call -send DTMF signal (for example 1234) -end call

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    ...would like to integrate certain features to it without vicidial installed it must work on freepbx We need features we can control through php scripts via AMI like : 1. adding sip users: 2. see user status 3. see Q status agents login succes rate etc like vicidial has 4. import and dial a campaign (that we have now) 5 automatic dialing for campaigns 6

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