Symbian sip software jobs

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    10,844 symbian sip software jobs found, pricing in USD

    ...is like this : IP 192.168.0.111.sip > 192.168.0.110.avt-profile-1: UDP, length 533 E`.1....@......o...n.......SIP/2.0 180 Ringing Via: SIP/2.0/UDP [login to view URL];branch=z9hG4bK18438941-9cbe-4ee8-b4f9-68f611bf3677;received=[login to view URL];rport=5004 From: "8111"<sip:8111@[login to view URL]>;tag=kencgqon To: "8112"<sip...

    $28 (Avg Bid)
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    3 bids

    I need you to develop some software for me. I would like this software to be developed for Windows using C or C++. Sip voip developers

    $502 (Avg Bid)
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    23 bids

    Simple sip phone or mobile dialer like galaxy sip settings to be installed on phone and being used for doing long distance calls. sure it has to be native and designed to company's desire.

    $19 / hr (Avg Bid)
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    44 bids

    I want someone who can make me register more than 1 user to receive call from my voip switch and send it to viber. i gave more than 300 viber accounts an...switch and send it to viber. i gave more than 300 viber accounts and i want to use them for termination. Project will get the developer good earnings despite the cost of the software or asterix.

    $514 (Avg Bid)
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    I need an application that runs on Android phones (4.0 or above) and 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. Full description below. Please review and message me with considerations. If any requirements

    $740 (Avg Bid)
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    20 bids
    FreeSwitch Kamailio SBC 4 days left
    VERIFIED

    ...I currently use Natpass, but am looking for a more open solution. You must have built out this configuration in the past. I do not want to pay you to learn. - SBC handles SIP Proxy - Registrations - Trunking - Security by locking down only domains we add I will need to see what experience you have before I hire you. I would like also to do an ongoing

    $61 / hr (Avg Bid)
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    1 bids
    Freepbx server setup 2 4 days left
    VERIFIED

    ...name Sales F). Extension 1006 having name Customer Care G). Extension 1007 having name Technical Support ii). Need to configure a sip trunk from Callcentric for incoming calls. iii). Need to configure a sip trunk from Localphone for incoming calls. iv). I have a GRANDSTREAM HT503 ATA that is needed to be configured with my free PBX server

    $30 (Avg Bid)
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    2 bids
    sip panel project 1 day left
    VERIFIED

    I need somewant to draw my house project in Dietrich's with optimatization of SIP panels.

    $138 (Avg Bid)
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    18 bids

    I am looking for someone, who can help me setup auto or a predictive dialer. Also, suggest a SIP partner.

    $249 (Avg Bid)
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    8 bids

    Compatibility requirements: Android / Android Tablet iPhone / iPad Windows macOS Nokia Symbian

    $1181 (Avg Bid)
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    15 bids

    ...where my SIP for external calling is configured now i need a configuration from Goautodial server to my external CRM. So I want sip trunk between GoAutodial and External crm so my CRM is able to make and receive calls using Goautodial and use line that is configured in GoAutodial ; Because my provider only gives me 1 node to connect the SIP so it will

    $185 (Avg Bid)
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    8 bids

    I have a voip Asterisk server that use SIP protocol to connect with the clients, normally I use sofphones, IP telephones or some App to make calls with this server. Now I need of implement a new feature in my website that is to make calls on-line in a web application, could only add a box with the field to number dialed that will make the call.

    $45 (Avg Bid)
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    5 bids

    ...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

    $1299 (Avg Bid)
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    26 bids
    Mikrotik Expert Ended
    VERIFIED

    Need Mikrotik Expert Now to assist with SIP Audio QoS Issue - RB2011UIAS-2HND-IN. Some minor cut out of audio. Looking for someone who can start now, access firewall via my Teamviewer.

    $134 (Avg Bid)
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    8 bids

    Me interesa crear un modulo para issabel pbx o asterisk para registro e ingreso de visitantes a conjuntos residenciales que soporte troncal sip

    $169 (Avg Bid)
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    we are looking a good freelancer or company who must be visit our office in Meerut (UP) and build a android based mobile softphone for us. for requierment please see attached file first.

    $445 (Avg Bid)
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    I have a CISCO 7970 VOIP phone which is running SCCP firmware. I need to chang it to a SIP firmware.

    $49 (Avg Bid)
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    I need you setup a freepbx with plivo sip trunk, outbound and inbound. Create 3 extensions, IVR with welcome message (2 languages) and main menu (2 languages) to a queue for that 3 extensions. The wav files are already created. Music on hold and position and waiting time information. Make that extensions can call by using the plivo trunk and receive

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    ...(IAX2/SIP) and vise versa. Requirements: 1- Work on any android mobile platform especially low price mobiles. 2- Register by SIP or IAX2 to any required server. 3- if the mobile have two SIM cards it should work for both and each by separate registration. 4- send calls from sip/iax2 to gsm and receive calls from gsm and send it to sip/iax2

    $1130 (Avg Bid)
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    11 bids

    Need Mikrotik Expert Now to assist with SIP Audio QoS Issue - RB2011UIAS-2HND-IN. Some minor cut out of audio. Looking for someone who can start now, access firewall via my Teamviewer. Thanks, Gavin

    $28 / hr (Avg Bid)
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    4 bids

    ...experienced with commercial setup and security hardening of VOIP servers. Below are details of setup: Project Description 1. Install Scalable Voip Platform Including Asterisk, a Sip Router a Billing System. 2. The installation will be on 2 VPS. Options include (Linode, Amazon EC, Digital Ocean, etc.) 3. Users can select and pay for DIDs online. DID selection

    $1481 (Avg Bid)
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    The goal to create with Asterisk video/audio conference room and connect it with jitsi clients. The Asterisk is installed on server. For the conference is conbridge module used. Jitsi should be installed on Raspberry Pi 3b+. After starting Raspberry Pi it should automatically dial the the conference room on Asterisk.

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    ...experienced developer with solid experience with XMPP and IOS and Andriod. The person/team must be capable of designing and programming a chat/talk application using XMPP and SIP. Preferred Skills: Erlang, XMPP, UI/UX Design, Graphic Design Preferred Database: Mnesia / MySQL /Postgres Important Functionalities: Real-time connectivity / Real-time

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    ...(GXW4104) for 3 PSTN lines. The Gateway is accessible but for some reason it stopped working for 1 week now. I tried different things but it's not dialing out. I added temporarly a SIP trunk so the systems continues to work. I need to get this Gateway working again. I can provide a Team Viewer access to a computer on the network and which has access to the

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    ...into the it: • Screen (either VGA or HDMI) • Wireless Keyboard and Mouse (USB Port) • Headphones (USB Port) • Power What must be installed on the machine? • Browser • SIP Client Who will use the machine? • The machine will be used by a call centre agent. • The call centre agent will open the browser. You must ensure that the browser ...

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    We require the assistance of a telecomms person to assist us in setting up a SIP incoming line with our provider DID logic. ([login to view URL])

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    I need someone to write me a very simple script that uses sip trunks (VOIP services) to call numbers from a database

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    SIP CALCULATOR: [login to view URL] HUMAN LIFE CALCULATOR [login to view URL] WEDDING PLAN CALCULATOR [login to view URL] CHILD EDUCATION

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    ...Developmentfor Android. Document ------------------------------- 1. user login by sip server user 2. show balance display 3. show call log 4. support incoming & Outgoing Call 5. Working Smooth 6. support g729,g729u & others voice code 7. Include any VPN Technology for SIP Block Country 8. Smart Graphical Interface 9. Include AdMob Ads **** When

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    Got a CP-7940G and i need it work with SIP the firmware need to change over to SIP

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    Trophy icon Design a Logo please Ended

    i need a logo designed for a company that organizes sip and paint art events. the name of the company is CREATE & PARTAKE. I possible i would like it to include something within the logo that gives it an international feel. maybe like a global behind the logo. Please make the logo artsy.

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    Guaranteed
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    100 entries

    ...freelacer or company who can build a mobile sip softphone for us. that softphone must have all call features and locally store call details on device like dial/missed/received call. Softphone must be able to support audio and vedio calls with suitable codec. G729 is must for this. Users will be able to add multiple SIP accounts. if you already developed

    $444 (Avg Bid)
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    8 bids

    getting sip error 408 when trying to connect to xlite for vicidial

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    Want someone to configure my elastix server for outbound calls. And add dial patterns. This is a simple job for someone with voip knowledge.

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    I am making a data system. I have all of the code and arduino files, I just nee...directories to put them in so they can communicate with one another, installing and importing older versions). I need to use PyQt4, pyserial, and pyqtgraph. But for pyqtgraph I need sip, and am having trouble with that. I am working on a Mac, running macOS Sierra10.12.6.

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    A guide on how to make SIP/VOIP http proxies for [login to view URL] and [login to view URL]

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    Hi We need to develop a SIP to Viber/Whatsapp gateway. The gateway should be able to pass voice calls incoming over SIP and forward them through Viber/whatsapp to complete the call to the called party number. The development platform/operating system is not important. The project should be completed either by using the Linux/Windows Viber/whatsapp

    $1300 (Avg Bid)
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    ...to send missed calls (outbound calls to be disconnected after 1-9 second of rings) - we do not mind if you work with any existing platform or configure any other available software likes IVM, Asterisk etc to the task as long it can fullfil the following requirements: 1. We are looking forward to send massive amount of missed calls using hundreds and

    $876 / hr (Avg Bid)
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    HI, I have a FreePBX up and running. Also, I get fax2e...IMAP are not activated. (We really don't want to activate it) (4) using different SIP trunk to send out fax according to the corresponding email addresses (5) We want one thing fix too. IAX2 extension doesn't have ringback when calling third parties on SIP trunk. SIP extensions are working fine.

    $246 (Avg Bid)
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    Few things that you need to do SIP applications are basically those applications using which you can call to mobile numbers through a network You need to do a little research about SIP calling You can find lots of free SIP calling credential from Zoiper for testing Below are the understanding of work for which you need to provide estimations Task

    $196 (Avg Bid)
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    [login to view URL] Extensive knowledge in iOS SDK (Software Development Kit)???- Solid knowledge in SIP- Solid knowledge in C programming- Solid knowledge in Objective C programming- Good knowledge in JavaScript- Good knowledge in media and codec handling- Good knowledge in HTTP and objective C

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    ONLY Knowledge with VOIP bid. SIP applications are basically those applications using which you can call to mobile numbers through a network You need to do a little research about SIP calling You can find lots of free SIP calling credential from Zoiper for testing Below are the understanding of work for which you need to provide estimations Task

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    ...two parts. One is mobile application and another one is registration server. The Mobile application will register to a server and accept call from that server with IXA or SIP protocol. After that call will terminate to a GSM network. (this part just like a traditional GMS getaway) This mobile application will work on only wifi Internet connectivity

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    ...Digital transformation and business strategy • Case study on a customer: I can provide content for you to structure an article with • Fall of ISDN in Australia and the rise of SIP Audio conferencing: • How to stay focused during company meetings (one point will need to be: using a high-quality audio conferencing unit) I need someone who has

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    Simple SIP redirect Routing system ,listen on port 5060 .It authenticate the incoming SIP invite source IP address , retrieve data by connecting to Mysql or Sql server, get values that should be appended to sip contact header edit/append sip contact : content ,then send it back to source of invite ,wait for ACK and send invite again if no ACK

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    I need a website designed. I need help designing my big cartel to make it look more professional

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    ...Asterisk must not bridge the final connected call. Asterisk must remove itself from the call-flow once the call has been transferred. The other pbx is currently setup under SIP and PJSIP with the same remote ip address. We are using the Asterisk pbx as an auto-attendant only. Our main pbx will send calls to the asterisk pbx and expects the calls

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    Hi, You previously worked on my FreePBX system. I would like to do a few things and could use some help. - Want to setup an extension that sends the calls to a SIP URL. I know this can be done in Asterisk but not sure how to do it in FreePBX - Perhaps it is time to upgrade my version, I am very behind. Largely things are working though. I am running

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    The software for create and management a video conferences by Web, SIP and h.323 protocol with supporting FullHD and UHD resolutions

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