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    8,372 flash media server sip jobs found, pricing in INR
    Country Sales Manager UK 4 days left
    VERIFIED

    ...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

    ₹304794 (Avg Bid)
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    7 bids

    ...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

    ₹308313 (Avg Bid)
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    7 bids
    LinPhone Development Project 3 days left
    VERIFIED

    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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    17 bids

    ...we take a server which run in static IP . all client should send calls to server IP from 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...

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    3 bids

    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.

    ₹7037 (Avg Bid)
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    9 bids

    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.

    ₹660 / hr (Avg Bid)
    Local
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    13 bids
    Freepbx and Asterisk settings 2 days left
    VERIFIED

    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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    4 bids

    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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    22 bids

    ...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 client where the IP PBX is. I also need to check

    ₹3299 (Avg Bid)
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    1 bids

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

    ₹3812 / hr (Avg Bid)
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    19 bids

    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

    ₹61281 (Avg Bid)
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    10 bids
    VOIP yate call pass 1 day left
    VERIFIED

    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

    ₹14661 (Avg Bid)
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    1 bids

    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

    ₹44128 (Avg Bid)
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    49 bids
    Wordpress Plugin for Phone Call 21 hours left
    VERIFIED

    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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    9 bids
    Actionscript 3.0 SIP 16 hours left
    VERIFIED

    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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    2 bids
    PHP SIP client 14 hours left
    VERIFIED

    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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    12 bids
    VOIP Project 11 hours 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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    14 bids

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

    ₹2492 / hr (Avg Bid)
    Local
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    3 bids

    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 [login to view URL] or yate or besip [login to view URL]

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    3 bids

    add sip trunk to elastix i have sip trunk from STC

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    I have 2 FreePBX Servers [A & B] Server 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 registere...

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    Looking to buy ready to use Call Centre CRM. Get back to me with demo. Complete documented 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 &bul...

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    9 bids

    we wish to have someone connect and configure our [login to view URL] to our SIP and Trunk

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    17 bids

    I need Linphone rebranded with our Company info, logo, colors and only allow the setup of SIP accounts over TLS. Removal of options to create a linphone account. This must be done on: IOS, Android, Windows and OSX The result must be packages ready to deploy through app stores. and source code must be handed over to us, as Linphone is open source

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    ...without ringing the phone. App will need to have standard features such as dashboards to see send data, user data, cost from carrier data. Must be able to connect to any SIP provider. Ringless Voicemail examples [login to view URL] [login to view URL] [login to view URL] https://www

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    Looking for a Tech Blogger to cover all the topics around VoIP, SIP trunking, DIDs, different countries legislations and Etc. Candidate needs to be familiar with the industry and love tech-savvy content Please when apply, send your samples

    ₹1979 / hr (Avg Bid)
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    18 bids

    We need to bridge standard SIP calls to/from our iOs/Android app written in Adobe AIR Actionscript. In other words: handle the RTP media part of SIP to/from spk/mic. If you don't know by now what is required, PLEASE DO NOT RESPOND! Fixed payment $200.

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    1 bids

    I’m throwing a sip and paint party. I’m looking for a artist to guide the class, to do a painting

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    20 bids

    I am unable to get calls from PSTN to freeswitch working. Calls from a SIP user into the switch (over the gateway) work. Calls between extensions and outbound calls work. I'm loading dialplan, configuration and directory with xml_curl (I had used a lua xml_handler script). I can confirm there is no problem on the carrier end...same carrier works with

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    I need a SIP client COMPLETELY written in ActionScript, so NO external libraries or other dependencies. It should be able to connect with a SIP server, ACCEPT calls only (so don't worry about dialing and invites) and handle that 2-way phone call (mic/speaker). That's it, nothing specific! If you know what you're doing, you don't need anything else

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    I require a voicemail d...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 [login to view URL] and sip.us. The application should feature the capacity to accept multiple channels for simultaneous dialing.

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    we are looking for en expert to develop a app that will act as sip gsm gateway i am including here a [login to view URL] to a software that was develop for window [login to view URL] 2, the actual software for window s i will drop it to you once we hire you 3. a suggestion

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    Hi, I need SMS gateway, Without using any third party services like twilio etc. I need my own gateway. Freelancer must have experience with Voip/PBX/SIP etc

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    6 bids

    We are looking person with specific experties SIP protcol PHP DontNet Also goo din graphics

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    Build a front-end with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar ...with REST API for Asterisk (login/off, register customers, password reminder) in Laravel, CakePHP or similar PHP framework, manipulate the Asterisk PBX from the interface (ivr, Sip/iax, did, ami/agi, voicemail, routes etc.)

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    43 bids

    ...SMS Gateway service that utilizes either PBX, SMPP, or VoIP - the platform must be able to run stand-alone meaning it can not use any 3rd party services such as a 3rd party sip provider, a 3rd party SMS service such as Twilio or Plivo. It must not need SIM Cards or modems. It must be able to create VoIP numbers (on it's own) ex; not using Google Voice

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    I have A Server and server have 2 Real IP IS ON USA i have a openwrt Router its Bangladesh and router have connected a VOIP DEVICE i want to send call THIS VOIP DEVICE now i have 2options 1) Install VPN on server and openwrt ROUTER and bridge network and set a static IP same serial on VOIP DEVICE useing MASQUERADE 2) make remote connections and USE

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    i want to setup an gsm voip gateway GUI using asterisk+a2b+chan dongle, use huawei e173u-1 and e153u-1. i already have a running server, but i want to setup a new server with the following features in a GUI: 1-sim card suspension detection based consecutive unsuccessful number of calls and discard(dongle stop now) the suspended ones as well as not

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    ...expert to fulfill the following: 1- Write a script for VICIDIAL installation with a single command line from [login to view URL] 2- Tutor the proper basic setup for interconnecting with SIP trunk 3- Tutor on how to use VICIDIAL in the following concepts: a- Press-1 campaign: where we will send a pre-recorded message that has an offer, if the customer is interested

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    XMPP-based client application integrated with a SIP-based client application.

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    We want someone to create a very simple and basic sip client app , we have a open source sip application which you can look at for reference , we need as simple as this . add sip authentication / dial screen

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    Need a browser extension or application for both windows and mac. The soft phone will require an agent to log in and be able to show when an ...from one agent to another agent. All call details will need to captured - Date & Time, Caller ID (Number) Start of Call and Finish, Agent ID Needs to be able to work across SIP and PBX providers using TAPI

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    ...interface to create extensions, reports, online and offline ext, and PBX functions FUNCTIONS: - queues; - reports; (with export to .pdf and .csv file) - IVR; - group capture ext; - SIP TRUNK configuration; - DAC; Mandatory: - send documentation from development; - log from customers and admins about changes; - each user have your color/logo (size from picture

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    Phase 1 Need to setup asteriskNow with ippbx (tata telecom). Setting up sip trunk. redirecting calls on specific numbers to specific sip no. At specific work hours or if specified through API, then to redirect calls to Mobile no. Setup recording of calls.

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    ...using Zoiper as softphone application to register extensions with IPPBX System. Our employees using Zoiper from inside and outside office. The protocols we are using are SIP and IAX We started a new office in Egypt, and would like to use Zoiper to register with same IPPBX System by employees working there. Unfortunately, its not working. We have

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    I need the Linphone client rebranded for Android/IOS and Windows/OSX. Simply with our logo / app-icon, removing options for creating or using linphone account, so only sip account is configurable. Furthermore changed to our colors. Packages for IOS and Android should be ready for deployment to the public stores, so we need the sourcecode when

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    Datalink International is seeking a skilled Android and iOS Application developer. Must have 5 + years of proven development skills. All A...developer. Must have 5 + years of proven development skills. All Apps developed will link over GSM Cellular networks to existing GPS and SMS Host Software. Must be familiar with SIP and serial data connections

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    ...that allow me to send calls to the mobile via SIP ( VoIP ) and this calls can me dialed ( Outbound ) Via the GSM ( the SIM card which is installed in the phone ) In more details Now we have a mobile device and Asterisk server, I want to link the mobile to the server and this server will send the call (SIP) and the mobile will make this call via the SIM

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