Sip configuration symbian jobs

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    14,599 sip configuration symbian jobs found, pricing in USD

    I need my website re-configured.I need you to design and build my online store. Need catalog creation with top mega menu with navigational features and banners Side bar menus with navigational menus Products linked with tags and types configured on smart collection relevant products to be imported from aliexpress and amazon using available apps within shopify, Edit the Title, Descriptions and ...

    $127 (Avg Bid)
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    3 bids

    I need some help with the basic configuration of my joomla template. Now it looks like [login to view URL] and at the end it should look like [login to view URL] (a long page with scrolling and dynamic elements).

    $165 (Avg Bid)
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    9 bids
    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

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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

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    Project Duration: 1 Hour ONLY Server Type: Linux OS: CentosOS Access Provided: Root SSH Issue: We keep getting 502 Bad Gateway NGINX error frequently on site/server. I want this fixed YOU MUST BE ABLE TO START WORK IMMEDIATELY 5* FEEDBACK WILL BE LEFT FOR YOU

    $17 / hr (Avg Bid)
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    ...giving issues and this issue needs to be resloved quick. We need a Magento 2.2.2 specialist and a server/database specialist combined. First need to check the server configuration and then resolve the issue that when a reindex is given not all items disappear from the site. Also the child categories don't show. See images of what is requiered. We

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    35 bids
    LinPhone Development Project 4 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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    18 bids

    ...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

    $723 (Avg Bid)
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    I have an Asterisk server for private use already running, and would like to acid...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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    Healthcare Software development company looking for someone to configure JIRA Desk Support project and Change management project

    $481 (Avg Bid)
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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.

    $9 / hr (Avg Bid)
    Local
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    13 bids
    Freepbx and Asterisk settings 3 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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    Work WPML setting configuration one website, Make it workable on my web. Professional in woocommerce web. We hope to display servial language.

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    Experienced server admin needed to review and correct cron configuration

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    6 bids

    I am looking for an experienced developer who has good PHP script & API knowledge. Also, developer must have the exp...developer who has good PHP script & API knowledge. Also, developer must have the experience with audio / video chat. I am using the AVchat plugin and I want to fix one configuration issue with plugin. Write above the bid (2+2 = ?)

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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

    $1126 (Avg Bid)
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    22 bids

    I have Configured SSL in my Tomcat, my application is Spring MVC. The SSL set up is not working on Server.

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    10 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

    $45 (Avg Bid)
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    Looking for someone to pass asterisk logs to, to determine why some calls are dropped or why calls are not routing properly.

    $52 / 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

    $836 (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

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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

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    49 bids

    Need to configure AWS WAF and Shield configuration on platform running with ELB

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    15 bids
    Wordpress Plugin for Phone Call 23 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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    Actionscript 3.0 SIP 18 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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    PHP SIP client 16 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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    nginx configuration 13 hours left

    No matter what the visitor types [login to view URL] or [login to view URL] or [login to view URL] i want all of the requests to be redirected to [login to view URL] currently i have managed to redirected requests that are coming from [login to view URL] and [login to view URL] to [login to view URL] but i can't find a solution for [login to view URL] to [login to view URL]

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    VOIP Project 12 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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    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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    Local
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    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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    add sip trunk to elastix i have sip trunk from STC

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

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    Hi dears My current loading time for my website is 13 second I need to be less than 3 second without ny effecting on the theme and current plugin. I'm using Woocommerce and this is my store [login to view URL] Thank you Saeed

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    Hi, i would like to cooperate with an expert on FreeSwitch configuration. Just to make it straight, I don't plan to use any web interface as a plugin extension like FusionPBX etc. I want configurations from the filesystem. Currently, I just have two questions. But the person that will help me on those items i plan to hire with a monthly consultation

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    hello, im looking some one talented, who can setup 123flash chat and make android and iphone application for mobil chat. only experiment guy's please!

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    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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    CONFIGURATION/HOSTING............................

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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

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    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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    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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    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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    We have haproxy running as our load balancer, and sticking sessions by port. Unfortunately, the queues are too big and users are unable to access backends, need to adjust configurations somehow.

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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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    Budget:20$,please do not bid more than 20$. You should have experience with Digital ocean,UBUNTU. 1)Server becomes slow at peak times ,need to configure nginx, hhvm,redis,load balancer. 2)Already site is running with nginx,hhvm,redis but slow. This is 1 hr job for expertise. 3)Server is having 8GB RAM 4)you can find out logs and make it faster so that server responds quickly.

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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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