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    9,910 sip tls srtp windows jobb har hittats, med prissättning USD

    Dear All, We are need a servic...4 asterisk machine and provide event logs and alert on the following status of the machine, IP phones, line and bandwidth below. - Server health - IP phone connection - SIP line - Bandwidth and latency of system to network - Connectivity Please submit your proposal by 25th of January 2019. Thank you, John

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    ...think, what if I could get a react.js coder who could create me a completely unique app that could access the Java REST API of Spark Wallet (running in server mode) and have TLS encryption end-to-end just like the Spark App? Is this possible? I will list the details of my current developer request below and you can tell me what is possible of a newly

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    Hi we are looking for the code of a webrtc button that can make a sip call(audio only)( [Logga in för att visa URL] maybe) be compatible with the mayor browsers at moment([Logga in för att visa URL]) and we should config stun and turn username and password sip server(asterisk) and number that will be called

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    ...VueJS, Bootstrap 4) 2. Back-end: (Python, Django, Django Rest Framework, API Development) 3. Database: (PostgreSQL) 4. Deployment: (Ubuntu 16.04 (Xenial), Let’s Encrypt (TLS), Docker)...

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    mail server for my website is definitely not secure (it is not supporting TLS). why the message is showing to my client if already install [Logga in för att visa URL] anybody know about [Logga in för att visa URL] clear me first then hire you.

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    Sip to native dialer app Features 1. this app need to be working in the background 2. the app file will have with all configuration user password to register to our sip server so client do not need to use user and password 3. the app register it self with a remote sip server 4. the result for

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    Android SIP Client Avslutades left

    SIP Client Application for Both IOS and Android prefer to go with Linphone Features : - Audio & HD video calls Multiple calls management (pause & resume) Call transfer Audio conferencing (merge calls into a conference) Instant Messaging with message delivery status (IMDN) Pictures and files sharing Contact list Call History Display of advanced call

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    I search someone that can fix problem with NopCommerce web payment plugins. You can check details on the photo. I take the site on...check details on the photo. I take the site on my localhost and it is working, but on the IIS server it has error. The error:"The request was aborted: Could not create SSL/TLS secure channel." I think the error in IIS.

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    ...DomainKey-Signature - List-Unsubscribe - List-Unsubscribe-Post: List-Unsubscribe=One-Click - Feedback-ID I follow all Gmail Guidelines and I have performed several checks, such ad SPF, DKIM, TLS = ALL ARE OK. I write to Gmail Support several times but they NEVER answer me. I need someone who is expert in IP/DOMAIN REPUTATION and EMAIL DELIVERABILITY. For any question

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    ...The features/ pages would include 15-30 Pages around. NAV of MF linking for fetching Price (NAV) info (we already have the membership of BSE). Retirement Planning Calculator SIP Calculator other calculators with various drop down Home page: Scroller images Corner: Login call us icon for: request a call back with preferred time slot & day 1. About us:

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    This project on creating Tools and Calculators for an Indian financial website. Examples of tools will be of Home Loan calculator, SIP calculator etc Please share relevant work references

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    ...com/OpenNMS/opennms) is an open source monitoring and systems management platform written in Java. It includes support to monitor SIP devices registered to and asterisk server and it does it by connecting to Asterisk's AMI interface and querying for SIP registrations. If the list of registrations change, OpenNMS generates an event or an alarm. OpenNMS depends on a

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    Good day, I am looking for a resource to configure queue metrics and freepbx, this is a new clean setup, new boxes. This you will start with clean setup. We have already a sip trunk provider and there accounts need to be setup also Both boxes is already loaded, we only nee to finish off the setup and configuration. We wish to extent both products

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    Free PBX inbound route Avslutades left

    If possible, I would like to route an inbound route based on TRUNK and NOT based on DID. FYI: My SIP provoider is NOT sending the DID, so the routing should go based on TRUNK (thus when an incomming call on trunk A comes, then it should be routed to extension 200, when an incomming call is comming on trunk B, it should be routed to extension 201 ect

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    Rebrand Chamilo or Moodle elearning system and Show installation and configuration of the complete system: BigBlueButton supports sharing of sl...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops.

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    i have a very basic client/server code working but it needs a custom coded tls handshake implemented in qtcpsocket manually to get the unencrypted text string. or get qssl to support the unencrypted mode setting [Logga in för att visa URL] small and compact easy to read no other external linked libs other than qt. only using

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    Hello! I am looking for a name and logo for my new painting party business. I offer mobile(no brick and mortar studio) paint and sip style parties mostly at local bars/restaurants and also private in home, but do not want to be limited to just canvas painting as I would like to host other craft classes as well in the future.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    ...sending messages • filtering messages • handling footnote • configuration of the webmail client with email account e.g. protocols (IMAP, SMTP, POP3) and certificates (SSL, TLS) and ports (e.g. 110/ 25 etc.) This is a basic and simple description of this module, we have shared a detailed requirement file which we were sent on chat. Technical requirements

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    AXNET VoIP Avslutades left

    I am Peter from Rwanda/ Africa. I am looking someone who can configure Skinny & SIP on our Cisco call manager express. We currently have E1 service which is active as well. Proposed Features: 1. Incoming and outgoing calls 2. Music on hold 3. Call transfer 4. Call park 4. Voicemail 5. Auto-attendant 6. Billing We have different brands of IP phones

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    I need to integrate incoming call and outgoing call in my website by using SIP library. Website must be developed by using ASP.NET MVC. I need an expert guy. I have sip server details. Only expert guy apply please. Open suggestion for js library but must support of my server details support.

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    Configure OPENSIPS Avslutades left

    ...an individual RTP media stream shall be as defined in the associated SDP description carried over the SIP signalling link. - The SIP signalling shall carry embedded ISUP signalling in accordance with the ITU-T SIP-I specification Q.1912.5. - The SIP signalling shall transport a 180 RINGING message without SDP parameters meaning the originator should

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    ...API 2. Call is initiated via Twilio Outbound Voice API using a Phone Number. 3. When call is answered, the call is forwarded to another Twilio phone number that uses Elastic SIP Trunk. It is attached to our IBM Voice Agent Service. 4.. Once Voice Agent is connected the user will speak to it., then be transferred to another phone number. We are looking

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    We have a Cisco AS5400 Universal Gateway which needs to be configured via an E1 ss7 . The Cisco gateway would interface with with TDM E1S and SIP Interface. We need to establish the Voice signalling via ISUP Controller which would be configured in the CISCO AS5400 via SLT interface via IP.

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    Rebrand Chamilo elearning system and Show installation and configuration of the complet...system and Show installation and configuration of the complete system: BigBlueButton supports sharing of slides (PDF and PPT), video, whiteboard, chat, voice (using Asterisk SIP installation), and desktops. Web Cam, Presentation, Integrated VoIP, Desktop Sharing

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    hi i have a ASTERISK SERVER and it have a SIP Account i have a kamailio5 installed on my local system now you need to register kamailio5 to Asterisk using SIP account when asterisk server send call to kamailio5 ==========kamailio5 have to send call to local gateway IP like [Logga in för att visa URL] if you can make it please bid

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    Dear freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want t...freelancer i want to install a VOIP Engine like opensip sems-server Kamailio you can install any one i just want to pas call 1st install Engine on openwrt router 2nd config SIP to pass calls codec g729 router model : TPLNIK 3420 v5

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    I have fusionpbx sip server working online. I need help to use cisco phones model 7945 with this pbx.

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    ...create SIP proxy to reduce Bandwidth now per call on SIP G729 Using 30kbps but i want to make it under 10KB we need to remove or change RTP payload or you can make a SIP client to rerouting call change PTIME or if you want you can do anything else as you want we have only a openwrt router in bangladesh and you can install small package like any SIP CLIENT

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    We're looking for a developer to join our team and help with daily tasks. Required skills: - SIP and VOIP in general - Asterisk - PHP & Codeigniter framework - SMPP - GIT Please share quick brief of your past experience related to skills listed above, your availability & time zone you're working in. Your written English must be good enough to understand

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    ...[Logga in för att visa URL] 1) Login Entry 2) Login Response 3) Normal Order Entry 4) Normal Order Entry Response 5) SIP Order Entry 6) SIP Order Entry Response 7) X-SIP Order Entry 8) X-SIP Order Entry Response 9) Spread Order Entry (Overnight) 10) Spread Order Entry Response (Overnight) 11) Switch Order Entry 12) Switch Order

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    Looking for help troubleshooting the setup for CartoDB on a production server with ssl/tls/ We have an Ansible script that gets everything set up based on [Logga in för att visa URL] but configured to run in production mode for ssh/tls/ The problem is there are various intermittent bugs that keep popping up and we need someone

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    ...IP Address (IP Tables)? -Disable PHPmyAdmin from External – Disable phpmyadmin running on all ports, 8080? 80? 8043? etc -Configure SSL/TLS servers to only use TLS 1.1 or TLS 1.2 if supported.<br/>Configure SSL/TLS servers to only support cipher suites that do not use block ciphers. Apply patches if available.<br/><br/>Note that additional con...

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    Application key features: * high definition audio and video calls * audio c...sharing * accounts reachable at any time, even if the app is closed (thanks to push notifications) * secure communication (encryption options) * compliant with a large number of SIP-compatible VoIP service providers allowing to reach everyone that has a "classic" phone line.

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    I need a Homer Seven server and ELK servers build to support SIP capture from Asterisk services. I need someone that works with SIP

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    FreePBX Setup Avslutades left

    We have 4 Digium D60 in office with a VPS that has FreePBX installed off site. There is VPN from phones to FreePBX. We are using Twilio for SIP. -Setting up extensions for all phones (4) -IVR -dial out rules -voicemail settings -inter office transfer setup -instructions on how to setup a fifth phone in the future (conference room phone) Start your

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    ...following error: 2019-01-08 01:07:11.651234 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;maddr={local_ip_v4};transport=udp,tcp) ATTEMPT 2 (RETRY IN 5 SEC) 2019-01-08 01:07:16.671190 [ERR] sofia.c:3146 Error Creating SIP UA for profile: default (sip:mod_sofia@{local_ip_v4} :5060;m

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    i need a software to remove or change SIP RTP PayLoad like if i install this software on os it have to remove or change all incoming and outgoing SIP RTP PayLoad if you can make it please let me know

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    Mobile VoIP Dialer Avslutades left

    Build a VoIP dialer app for Android/IOS and WM using either Linphone or Csi...Account auto provision/Signup via REST. - Account Balance Check Via Rest. - Account Topup via rest.(In App Payments) - Multiple Sip registrar options (Selectable) - User Credit display in app. - Voice/Video/Chat support - *** SIP Header Compression support! within the app.

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    ...and can’t sent us any email. We think that this problem is coming from different elements. SPF and dns setting in our primary dns servers, lake of SSL certificate in Cgpro (TLS), CGpro wrong or incomplete relaying settings, maybe something else? Our environment : CGpro (Stalker communigate pro server) on linux [Logga in för att visa URL] is a secondary dns server

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    solución pbx Avslutades left

    Tengo una centralita pbx con issabel y 2 troncales sip con gateway goip en redes difefentes pero siempre se desconectan y tengo que reiniciar el servidos pbx para que se vuelvan a restablecer

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    Design project Avslutades left

    Sip and paint artist needed for one week event

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    need hobby geeks Avslutades left

    ...Customer Relations/Professionalism 5. Speak clear English 6. Basic Computer Knowledge 7. Have a laptop or desktop with windows 7 or higher. 8. Photo ID 9. Paypal account for payments 10. Headset Optional but preferred. for windows sip phone, or android, iphone 11. willing to be creative, and outside the box thinker 12. Honest Contact me to Learn More

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    Hi Dear In running local FreePbx server under elastix and the server link to Sip line from local Telecome company and there is another network card for internet connection My local server is online now via port forward from the router and it’s working with Zoiper app very fine in computer an mobile... But the problem its just working with Zoiper nothing

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    I am looking for someone who can build a sip calling APP using a Mobile Hybrid platform such as ionic or phonegap or any other Mobile Hybrid platform. the requirement is to connect to our SIP servers and make calls through a SIP proxy. we will provide all the server details needed

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    Description. A simple PBX pure software would have to be [Logga in för att visa URL] software should be installed on Windows Caller calls the number to Will and can select appropriate department through IVR. The language should be stored under MP3. Customer lists to maintenance music + and waiting position. Once customer is connected with the agent and the agent wants

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    ...a html ''Coming Soon" website template. The site has a contact form. It uses the old send mail via php but that doesn't work with my dedicated hosting. (Microsoft Azure, Windows IIS platform) phpMailer works on this host (have another developed site where it has been working on the same host) I want phpMailer to be implemented for sending emails via

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    I need a website which is based on wordpress for hiring people. The website have to be using SSL/TLS certificate. People should be create account on my page, then confirm their mail through the own mailserver. Then they can write with me through ticket system and upload their documents (e.g. CV, photo etc...) I should be can edit the member within

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