Configuration redirect server sip jobs

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    16,037 configuration redirect server sip jobs found, pricing in INR

    ...done allowing for payment via credit card, debit card or EFT. 6) Further configuration and setup must include various plugins/extensions to be bought by my company for integration into the system such as Events, Case, Contribute, Mail and Report related extensions. 7) Configuration must also include different types of subscriptions for members and the

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    We need to move from a blog on a subdomain to main site and therefore need to put in place 301 redirect in .htaccess file. The difficulty is that there are accents on some of the source URL's

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    I need a php redirect system made.

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    Healthcare Software development company looking for someone to configure JIRA Desk Support project and Change management project

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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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    Local
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    Freepbx and Asterisk settings 6 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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    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

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    SSL Configuration Bug Fixes 5 days left
    VERIFIED

    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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    ...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

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    New SSL certificate purchased and all ok on hosting server. Have been advised CRE loaded code needs to be changed.

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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.

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    I have a wordpress website that has a specific user that is limited on what they can see on the backend. For that user, when they login through /wp-admin I'd like it to go directly to Woocommerce rather than to the Profile tab. Should be very quick and easy for someone who knows where to find this on the backend.

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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

    ₹70548 (Avg Bid)
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    7 bids
    VOIP yate call pass 4 days 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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    app development 4 days 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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    Need to configure AWS WAF and Shield configuration on platform running with ELB

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    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 4 days 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 4 days 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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    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 3 days 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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    Create SNOM Phone Dial Plan 3 days left
    VERIFIED

    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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    After moving a website to a development subdomain, the site is now constantly redirecting to [domain]/[login to view URL][domain]. I require someone to resolve this asap.

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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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    Redirect Single php Page 2 days left
    VERIFIED

    Have Redirect Single php Page to another domain single url.

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

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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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    change the redirection link and make it functionally work

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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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    ...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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    I'm looking for a Graphic designer who can layout stunning Magazines, create custom ads for clients and is also familiar with php, html, wordpress for website design/ maintance.

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    ...highly skilled WordPress professional with strong background in web security/hacks and malware removal. Our website [login to view URL] currently shows signs of a malware redirect when trying to access the homepage it periodically sends visitors to other spam sites a sign of malware. We installed the plugin WordFence and scanned the site to find some

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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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    the project is to build ionic app plugin out of linphone sip framework or linphone-cordova-plugin. Plugin should have following features. Audio calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Call History Echo Cancellation Call quality indicator Secure user authentication : md5 / SHA256

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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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    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 b...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 have an old website we want to move to a new domain name. We have new website done and want to turn off old website but don't want to loose any traffic. The [login to view URL] needs to be setup to divert both the home page all old pages. If you can prove it is working on home page and one more linkl I will fill in rest myself. I had tried but have a problem with https:// causing issue wh...

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    ...web scrapping. When turn on web browser to check and visit a website, may be moved from the redirection page to the main page. i want to get the cookies that occur on the redirect page when the page is redirected. Programmatic web browsing is required to navigate to the last page. If we use a web browser, the redirection is automatic. I want something

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