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    12,820 application android sip voip робіт знайдено, ціни вказані в USD

    This project is to deliver recorded video how to use cordova-plugin-sip plugin in ionic 3

    $196 (Avg Bid)
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    I need to create and confing server. i need to forward calls to it and to see the header and make tests can be done on linuxcpanelwinvps

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    I need someone to configure a C2901 router that is running CME. The router is partially configured but still some issues that need to be worked out. - Outbound delay - Dialpeers - auto attendant

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    VOIP demo callKit Закінчився left

    I need an iOS VOIP demo using callKit and written in swift 4. it should handle making calls, receiving calls, connecting calls, and terminating calls.

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    Ho la necessita di registrare un telefono cisco cp-6945 su un centralino grandstream ucm6204. il telefono ha già caricato il fw sip. Il telefono non permette la configurazione tramite web access ma solo tramite file XML

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    I need to develop an Android Studio application that allows to include the SIP / VOIP SDK to connect to ASTERISK The project must: 1.- Connect via SIP to an Asterisk telephone exchange 2.- Call from a number console 3.- Configure (add / modify / delete) SIP accounts 5.- Auto Answer Caller More details (ask) ////////////////////////////////// Necesito

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    voip freepbx Закінчився left

    i need guy who has core kownledge in freepbx

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    voip calling app Закінчився left

    I need a responsive website. I would like it designed and built. voip calling app which generates a unique virtual number for the user instead of showing his/her number. person has to first register to get the unique virtual number from which they will call.

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    VOIP calling app Закінчився left

    I need an Android app. I would like it designed and built. VOIP calling app.

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    Voicemail Drop Windows Software Закінчився left

    ... The application will feature voicemail detection function where an outbound call is placed from a csv list and is directly router to the recipient's voicemail without ringing. Afterwards a pre recorded message is automatically left. For placing calls I would like to use SIP trunking or VoIP such as [увійдіть, щоб побачити URL] and sip.us. The application should

    $175 (Avg Bid)
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    VoIP Business Startup & Web Portal Закінчився left

    Hi, I need training on VoIP Business asterisk , Mysql , Billing Platform and security.. Linux asterisk training Learning about SIP Also i am looking to set the business in uk . Below are competitor websites i need similar to this setting up [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL] [увійдіть, щоб побачити URL]

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    Voip asterisk administrator expert Закінчився left

    i need to setup a complex voip system, need to purshase a telephone number from [увійдіть, щоб побачити URL] (which is not a local provider) and configure it for incoming and outgoing calls, you need to know lots about routing, area codes and prefixes, since i cant get local dids and need to get the did's from international voip provider.

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    ...need of a customization of the opensource VoIP / SIP client LinPhone or Ring or Wire (if not than Jitsi). Preferably LinPhone, but if not possible one of these. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization

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    Questions About Voip Закінчився left

    I'm looking to get some guidance on a voip project.

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    So I need to hold a call and check if exist on a whitelist or not then pass the call or drop it. for any incoming call we need to check the number 1 hold call on wait for 3 sec lookat whitelist if pass job is done if not generate a call on a specific route and look at the feedback if recived call reject then reject the call else disconnect check call and unhold the original call and pass it to ...

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    install and configure freepbx Закінчився left

    need freepbx install on an inhouse server. have sip trunk and pstn lines. need configuration for IVR. 4 auto-attendants. simple call flow. need call monitoing, call recording and SMS

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    Gsm voip gatway Закінчився left

    I need an Android app. I would like it designed And Build Android application make mobile phone act as GSM to voip gateway To connect with Asterisk PBX We needs to use an android cellphone as an asterisk channel/Gateway. His aim is to make phone calls using his Asterisk PBX through a cellphone running android. It aim to make phone calls using

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    Country Sales Manager UK Закінчився left

    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. We are looking for a

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    ...end of the year, we are going to launch a SaaS in the UK. Target group are delivery restaurants without website that allows ordering. We provide leads for a start, dedicated SIP phone numbers, a CRM, and training. We offer high sales commissions and contract extension commissions. We expect impeccable UK accent - no exceptions. When you apply make sure

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    VoIP telephony Закінчився left

    We have iPECS-100 behind Chekpoint firewall. All customers use soft phones clients. Have periodically experience communication problems. Need a person for a one-time or permanent job

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    LinPhone Development Project Закінчився left

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    ...his SIP server and we need to create separate account for each gateway . and calls should send to specific termination . call should pass with sip , g729 or g723. in client end we want to setup asterisk module which can convert calls from sip to sip and pass the calls to gateway . 01. asterisk or SBO server. which receive calls from many sip server

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    I have an Asterisk server for private use already running, and would like to acidify a Trunk using an FXO VoIP Gateway, for this is necessary to create a sip trunk in Asterisk and I do not know how to do. In my attempts, I can even connect SIP between them, but I can not complete calls.

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    android voip call application Закінчився left

    System requirements Registra...and database mysql Through the application you can make a voice call between registered users on the system The user is activated after registration through the database The IP number, MAC address, device type and user name are stored in the database The call is secure and encrypted between the parties Simple application

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    I need a server administrator with experience on VoIP technology. FreeBPX, Asterisk, SIP Clients Cisco SIP phone provisioning and some other SIP phones. Developer most commute to office in Rupnaghar India. If you don't leave in India please don't apply.

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    Місцевий
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    Freepbx and Asterisk settings Закінчився left

    I am using for our office a pbx with regular sip phones and some softphones. The softphones work using Bria Mobile with Push notifications enabled. However, the phone doesn’t work properly on background meaning that push notifications don’t work as expected for receiving calls. Here is the page where Bria mobile explains the settings that needs to be

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    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

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    Project for Anton F. Закінчився left

    ...I have a IP PBX with 2 sets of 10 numbers. Set nr1: comes through normal ISDN to IP PBX Set nr2: comes through incoming Vitual Voip DID. All the numbers of set nr 2 work but i have 1 nr that does not work as i get error sip 503. For security reasons , you will be operating through Teamviewer on my computer which is connected to the remote site of my

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    i have a Linux server its running on Apache tomcat i have some .jsp file on webapp forder its an api for vos3000 like i want to insert update delete but i dont know how to do you have to fixed it check the attach file here is all info i dont release any fund without test

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    asterisk SIP tracing and engineering Закінчився left

    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    $53 / hr (Avg Bid)
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    21 заявки

    We are in a need of a customization of the opensource VoIP / SIP client LinPhone. The development has to be done in two phases: Phase 1: Customization of mobile + tablet apps (iOS + Android) // Objective C = iOS + Java = Android --> ready in two weeks. Phase 2: Customization of PC apps (Windows, Mac OSX and Linux) // Qt/QML = Windows, Mac OSX and Linux

    $836 (Avg Bid)
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    10 заявки
    VOIP yate call pass Закінчився left

    i need to install yate on openwrt and pass calls server to my gateway we pass call useing sip to sip if you can make it please bid

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    app development Закінчився left

    Android sip cellular gateway we are looking for an expert in developing mobile app to develop an app that will expect voice calls using usip server and dial out local number using the mobile network

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    VOIP phone system Закінчився left

    we need a phone system based on twilio platform.

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    Wordpress Plugin for Phone Call Закінчився left

    Hi, I need a wordpress plugin which will allow the users to place a call on the website. I will be integrating a sip gateway for same. Users can place free calls with some restrictions like 1 minutes, or a 10 seconds ad after every 1 minute. Paying users can place unlimited calls until their credits are exhausted I wish to achieve a website like https://ievaphone

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    Actionscript 3.0 SIP Закінчився left

    1. REGISTER + SUBSCRIBE to SIP server, with authentication 2. Accept INVITES 3. play inbound SIP packets and convert microphone to outbound sip That's it. If you've done it before, it's 1 hour work. Fixed payment $100. We need Actionscript in AIR.

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    PHP SIP client Закінчився left

    PHP SIP client: 1. REGISTER + SUBSCRIBE to SIP server, WITH authentication 2. Accept INVITEs to send only, NO recv (so no traffic inbound) 3. Return WAV file to caller; depending on called number either file 1 or file 2 4. BYE to disconnect, that's it. - FIXED $40 for working code (do NOT ASK FOR MORE) - It's not even 30 minutes work if you've done

    PHP
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    VOIP Project Закінчився left

    Signalling work with CISCO CUCM Understanding VOIP - SIP (including Blind and Attended Transfer implementation) - VOIP Call analysis – Wire Shark or similar - Visual Basic Script language - Proprietary Cisco SIP protocol extensions - Cisco CUCM - Call flows of the Attended Call Transfer - Cisco Finesse handling of the Call Transfer Budget as outlined

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    Requirement two-way audio/video SIP based communications form the mobile phones to Asterisks on ARM based targets and web browser targets

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    Місцевий
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    Create SNOM Phone Dial Plan Закінчився left

    I am looking for a SNOM phone guru that can create a dial plan for me with the following...strip the '"9". A long distance call would be 91XXXXXXXXXX. We do not want to use time outs for anything but the international calling. We require the complete string for SIP account 1, 2, & 3, Delivery in a text file. PBX: 3CX Professional Phone: SNOM D765

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    i have voip device and i have small router it have 32mb ram os Openwrt i want to run 32calls so you need to use SIP2SIP register USA SIP server >>Bangladesh Openwrt Router>>>VOIP Device we cant use asterisk cz its need too much space i think best libre [увійдіть, щоб побачити URL] or yate or besip [увійдіть, щоб побачити URL]

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    Integrazione api voip Закінчився left

    Vorrei integrare un sistema VoIP nel mio sito web così cliente possono vedere la loro fattura e possiamo vedere i loro menù.

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    add sip trunk to elastix Закінчився left

    add sip trunk to elastix i have sip trunk from STC

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    Paid VOIP services app Закінчився left

    Application that allows people to instantly phone a specialist, on a pay-per-minute basis, to receive recommendations

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    ...A: On Cloud And SIP Trunk has been created SSL is configured by "Let's Encrypt" Server B: On Local And SIP Trunk has been configured SSL is configured by "Self Signed" Result: Server A - SIP Trunk is appeared as not registered Server B - SIP Trunk is appeared as registered! I need to get all Registered, and the call is go through SIP...

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    ...installation and setup guide will also be required. I need you to develop some software for me. I would like this software to be developed for Linux using PHP, with Asterisk voip for call center. Solutions - Features - Call Features • Call Detail Records • Call Forward • Call Monitoring • Call Parking • Call Queuing • Call Recording • Call Transfer &bu...

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    Raspberry pi as VoIP GSM gateway Закінчився left

    I want to use raspi as gsm gateway. This is the basic task. Details will be explained.

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    Independent dialer system Закінчився left

    we wish to have someone connect and configure our [увійдіть, щоб побачити URL] to our SIP and Trunk

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    I have an iOS app which uses Sinch api for the VoIP calls, i'd like this replaced with an opensource solution, such as FreeSWITCH. The app uses usernames and not phone number. It's in Obj-C, with php services, mysql DB, hosted on Amazon AWS. I expect excellent clear quality calls.

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    SIP面板中找到100家从事房屋建筑的公司 如果事情不明确,请写信给我,我会尽力解释。 在中文里,我通过谷歌翻译写作。 我的母语是俄语。

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