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    11,175 sip tunneling htc jobs found, pricing in CAD
    FreePBX install. 6 days left
    VERIFIED

    In freePBX, I need the following setup: a.) I will supply SIP trunk with 2 DIDs. b.) Create a queue, and 10 extensions that will be routed with the agents evenly who ever has less work. c.) Record all phone calls, and have the ability to pull up reports by extension and FOP2 panel. d.) Voice mail needs to be sent to email, indicating the queue the DID that had the incoming call. e.) When ringi...

    $39 (Avg Bid)
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    1 bids

    Prior Experience of: - Location: Bangalore - 3-5 years Experience - Writing augmented reality code for iOS (Swift), Android (Java), and Unity (C#). - Designing client-server architectures to handle thousands of users worldwide. - Build 3D VR experiences with unity or unreal game engine - Knowledge of Unity Scripting, Engine and UI. - Work on different VR Kits like Oculus, HTC Vive etc. - Implem...

    $37 / hr (Avg Bid)
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    Trunk is getting [login to view URL] needs to be configured [login to view URL] to terminate on Fixed DIDs

    $30 (Avg Bid)
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    5 bids
    Android App / Gateway 5 days left
    VERIFIED

    We are looking for a developer to build up a android app. App will make a connection between an incoming VOIP calls to the app and outgoing SIP calls from the simcard in the phone. Basically, an internet call will come to the app with targeting a phone number, than app will call that number with using the pysical simcard in the phone, and make an instant connection between incoming internet call...

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    1- Creating a SIP trunk between 3CX and Panasonic TDE-100 2- The ability to make calls internal calls between extensions behind the 3CX and the TDE-100 P.S: 3CX and the TDE-100 are not on the same network We succeeded to create the Trunk and get it in service and to make calls from the TDE-100 to the 3CX but the other way around is not working! We are getting a response "Not Found"

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    Create a service like [login to view URL] We need: 1) Our clients calls from Telegram app to our Telegram account should go to our Asterisk PBX via SIP 2) Be able to from our Asterisk PBX call our clients Telegram accounts. 3) Handle 50 concurrent calls Open source example:P roject [login to view URL] Here you can see how the bot works [login to view URL] [login to view URL] Libraries use...

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    VR App Development 5 days left
    VERIFIED

    Hello Freelancers, The idea is that I want to create opportunity for people who can't visit Disney so I want them to experience the different rides through Virtual Reality. I am looking for Game developer to develop VR App(Roller coaster type rides). You need to design models for Disney rides and should be able to deliver for at-least two of these(Android App, HTC VIVE or Oculus) platform. ...

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    11 bids

    Want to convert avaya 1608-l to sip so that it can be used on asterisk pbx

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    Create a service like [login to view URL] option to use their open source project [login to view URL] so that our clients can call our asterisk PBX from their telegram accounts. Here you can see how the bot works [login to view URL] [login to view URL] Libraries used: [login to view URL] [login to view URL] [login to view URL] Supported call types are: SIP->TG call by username SIP->TG...

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    3 bids
    Xamarin voip/sip call , 4 days left
    VERIFIED

    I need audio/video calling p2p on Xamarin Cross Platform. Only one-to-one communication no conference needed. Audio and Video calling between app to app.

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    I need audio/video calling p2p on Xamarin Cross Platform. Only one-to-one communication no conference needed. Audio and Video calling between app to app.

    $715 (Avg Bid)
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    10 bids

    Configure and setup 3CX PBX in windows10 using twilio sip trucks.

    $65 / hr (Avg Bid)
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    13 bids

    Please read careful before replying: Case: - Existing Freepbx; up and running! - Existing SIP account successfully registered on MicroSIP application on Windows machine; thus the SIP account works 100% fine using MicroSIP. > By this I mean: I received SIP information from my phone company. Using this SIP information I can place outgoing and incoming calls using MicroSip. - The SIP account is ...

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    We need to set up following things in FreePBX 1. SIP Trunk - 1 2. SIP Extension - 1 3. API For Fetching Current Call 4. API For Fetching Call Recordings 3. API For Making Outgoing Call

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    We are looking for some Asterisk work on a number of items. We are not using freepbx or other GUI. The current tasks consist of the following: The configuration is two SIP trunks from Twillio. One trunk works the other does not. Need to troubleshoot. All calls coming into the default extension of either of the trunks must simul ring to all extensions existing now and any that are added in the...

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    I need a logo designed. Serve Impact Prosper SIP I would also like the logo put on (2) banners. (1) with dimensions for my Facebook Page & (1) with dimensions for my Facebook Group. I've attached an image of the color I'd like to use.

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    5 entries

    i need training on Voip i am appearing for a voip engineer job to crack interview i need training on network automation ,voip(Sip,rtc,webrtc,etc), managing network enterprise if i crack the interview i will give you back something

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    i need training on Voip i am appearing for a voip engineer job to crack interview i need training on network automation ,voip(Sip,rtc,webrtc,etc), managing network enterprise if i crack the interview i will give you back something but 3000 indian ruppee

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    Personal Site For SME Features have 3 in 1 module - Accounting + Invoicing + HR & Payroll this for only Malaysian enviroment only. Finally this can be have login for admin, finance, hr, sales, and staff only. ** this site for personal used so don't have any feature for subscribe or other can use this site. So only have backend login and front end for display the general information....

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    Hi Ram i have two asterisk conmected via IAX, when call is NO ANSWER or BUSY or CANCEL im not getting good Sip cause but 503, exemple client send call he get ringing 180 than after timeout (no answer) he get 503 instead of 408, the gateway is sending correct sip cause to 1st asterisk but always congestion(503) except when call Answered I think something wrong in [login to view URL] or [login to...

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    A short VR interactive experience to be developed on HTC Vive. The experience is similar to Richie's plank . Will have use HTC Vive Trackers to have interaction with the snake in real world. The experience should be of good quality, the visuals should be realistic. The aim of the game to make the player feel the "presence/immersion " as much as possible . The 3D assets (Snakes, croc...

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    Allowing live audio group chats for up to 40 peer devices to receive broadcast audio from a master device over a local Wi-Fi network. Allowing one device (the “master”) to broadcast live audio to multiple “peer” devices over local WiFi. The app should be built cross platform in Flutter, to allow for voice to be broadcasted between Android and iOS devices running on the s...

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    I need small businesses to find my link when they need Business Internet, Business SmartOffice, Business TV, Business Phone, Business Trunks, Business SIP Trucks, Business Voice Edge Select, Business Voice Edge, Business Ethernet Service, Business SD-WAN products and/or services. The link is a jotform that asks some simple questions about their company, what products they are looking for, their co...

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    When making a outbound call from Skype for Business softphone unable to hear the ringing on SFB app. But the call rings at the other end and when answer the call works fine. Current setup is when calling from SFB has a trunk connecting to Asterisk freepbx and the SIP trunk to the provider is configured in Asterisk. it would be great to have someone who has knowledge in both Asterisk and SFB serve...

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    Connecting WebRTC compliant devices to the SBC WebRTC Gateway at the edge of the cloud. 2 MUST conditions for solutions are 1. IP device with WebRTC support SIP over WebSocket signalling 2. RTP media transfer, communication protection as they cross network boundaries Embedded device environment: Embedded Linux Libs Available open source libraries can be used Solutions must be efficient, whereve...

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    We are looking for B2B sales agents to generate qualified leads of companies interested in upgrading their current VOIP system, saving money on their current VOIP system or also companies looking for a new VOIP system. We are most advanced VOIP company in US with competitive price. Criteria: Willing to move to a cloud based VoIP phone system We provide US and Canada numbers only. Minimum spe...

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    Interconnection between WINDOWS SOFTSWITCH that has PUBLIC IP, and Asterisk PBX (DYNAMIC IP),, We want to connect the Asterisk PBX as a trunk to the windows softswitch so calls from Softswitch to Asterisk PBX can be successfully connected.. Please note: DYNAMIC DNS is not preferred.. Thank you.

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    any one can develop like that voip sip dialer! working on voip switch demo dialer [login to view URL]

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    Hi, we are looking anyone who can create web interface for asterisk where we done below task 1. add / remove sip trunk 2. check sip trunk status 3. realtime calling (live calls) 4. force/terminate calls 5. generate calls from mysql and send CDR to any Mysql 6. generate TTS calls from mysql and send CDR to mysql 7. Capture User Input (DTMF) and send to mysql along with CDR thanks

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    A short VR interactive experience to be developed on HTC Vive. The experience is similar to Richie's plank , we Will have use HTc Vive Trackers to have interaction with the snake in real world. The experience should be of good quality, the visuals should be realistic. The aim of the game to make the player feel the "presence/immersion " as much as possible . The 3D assets (Snakes, c...

    $2228 (Avg Bid)
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    We have Two Avaya SBC Servers with Two SIP lines and 2 IPO Office Ver11 Server edition. We need configure 2 SIP trunks on SBC and Its should work with IPO. Also, We need configure Avaya XI Workspace with TLS and SRTP with Public Access.

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    I need an sip voip application that runs on Android phone (4.0). the app act as sip voip gateway termination device. It can receive calls from sip server (registerd with sip softswitch) and forward it to gsm network. The application should then forward the audio and convert from SIP to GSM and vice versa. Full description below. General Deliveries The application working in APK format Full sour...

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    I would like to have a light sip/voip server, web interface/dashboard for manage user/group/company/geolocation of users, and android app push to talk. For more convenience please take a look to [login to view URL] or [login to view URL] NO website...backend+web+android app

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    I would like to create a VoIP Android App similarly to an existing one: Imsdroid from Duobango. For the beginning I would like to keep it simple to have both voice and video capabilities, message functionality is not that important (it could not be included). Also i want this app to be able to include a different agenda, from the one existing on a phone, something like a database which will conta...

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    Hello, I'm looking for someone who can setup my 4FXO to make outgoing calls only, from an external SIP software (SIP registration from another IP connection, no local uses). 3 telecom providers will be connected to this VoIP ATA. The device has been bought to ChinaSkyline, it looks like this : [login to view URL] (but with 4 FXO ports). I give you a TV code for your work. Thank you for you co...

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    Hi, want a banner for LinkedIn. I graduated in a course for HTC and want to say it. I want something to represent high ticket sales and say HTC

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    Hello, I am in need of an engineer to setup FreePBX Server (PBXact) remotely. The following items are needed to consider project completed: 1) SIP Station Trunk Setup for all outbound calls. 2) Second SIP Trunk for Inbound Calls. 3) Configure for remote phones utilizing Sangoma Auto Provision. 4) Configure Firewall and protect system. Looking for someone to start on this project now. Thanks,

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    Hi, I need assistance with a voip migration project, you must have prior experience with 1) Migration from Asterisk to freeswtich 3.6 or 4.0 Multitenant enviroment 2) Setting up FreeSBC / Pro SBC [login to view URL] 3) Dialplan setup 4) SIP Trunk setup. 5) Virtualized setup on Vmware or AWS. Project must be completed in 7-10 days.

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    We need the Business case or proposal for call center setup to submitted to telecom Companies we need the partner ship with telecom operators: Details: Over all Call center Setup details included SIP over Point to Point Getting Golden Number for SIP Installation and configuration survey and contract on behalf of the telecom Companies more detials will be provide via chat.

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    We are working in Voip terminations, and VOIP wholesale I have a SIP terminator but I can't interconnect with him or reach his SIP without using Cisco VPN, the VPN protocol of communication is called IPSEC + xauth I tried the VPN connection and it connects, but then I lose access to my SWTICH. What we need to do is : Initiate succesfful connection with Our SIP terminator END point using VPN...

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    I have opensips installed with 4 Freeswitch servers, IP authentication is working fine but in SIP authentication (i.e., Register) i need to append following code. # Setting accountcode in P-Accountcode variable remove_hf("P-Accountcode"); remove_hf("P-Pricelist_id"); cache_fetch("local","accountcode_$fU",$avp(accountcode)); ...

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    We would like to work with a developer to develop a minimum viable product (MVP) / Proof of Concept for an RTP recording system that can accomodate multiple codecs (G.711 / G.729). The application must run on Linux and must be configurable as a service/daemon so that it starts automatically at power up. The initial version will be simple to test the concept and functionality - it will receiv...

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    We need to create SIP compatible softphone for our asterisk server need to include: push notification transfer and conference BLF buttons if possible auto provision template

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    I need an unity 3d vr game developed with multiplayer for pc HTC vive. We expect realistic quality and 1 map environment. This will be for a complete game development. This is a multiplayer htc vive fight. 5 players. 1 player is like a God with abilities. The other 4 are super heros each with abilities (unknown at this time what powers are going to be) Will fight the God like giant (he will dwarf ...

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    We are looking for an asterisk / full stack developer who can help us build a call center management platform platform and dialer using Asterisk. Here is the basic outline of the project. This project will require the following. Agent Control (Manage agent and customer interactions) Call Routing Queue monitoring and reporting Provide call information (referred to as call pop-up) to agents before...

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    We want to make VOIP Calling Card Website for Our International Customer, Rate plan fetched from sip server. Client can Signup from Our Website and Get Username/ Password Automatically from SIP Server using API. Database/mysql required. Also intergret Paypal, Visa Mastercard amex Card etc. for buying credit using calls from Dialer. [login to view URL] (demo) freelancer come to visit website ...

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    I need to use an android cellphone as an asterisk channel/Gateway. This will make phone calls using Asterisk PBX through a cellphone running android. In this case, Android phone will be acting as a GSM Gateway for Asterisk. The application working in APK format Full source code Simple manual for compiling and generating the application from source Features : -Route call from SIP to G...

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    I am looking for a detailed instruction for the installation of Opensips, Opensips-cp, freeswitch, homer, cdr/billing(eg cgrates with gui), all in a single machine (real or VM). The host system would be Debian 10 64bit. The documentation should be taken from a novice engineer and get reproduced in a step by step manner. Everything would be in LAN and the IP would be [login to view URL] with gatewa...

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    I need assistance in setting up my FreePBX home server. I'm using Cisco WebEx Teams and want to use the call in feature using the SIP URI provided by Cisco (example sip URI: <extension>@[login to view URL]). So I've set up an extension with the DIAL property pointing to that specific SIP URI (which means that I'm not dialing the SIP URI from an internal endpoint). The problem...

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    We need to configure sip trunking and voip

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