...la configuration ( le découpage réseau avec explication ) il faut interdire l'accès au serveur aux pc dans vlan student et juste les vlan prof peut accéder au serveur un nat ipv6 RIP avec loopback...
Sur un Fortigate 300A en production, vous devez créer une règle de routage NAT. Il faut utiliser l'adresse IPv4 public déjà configurer et lui ajouter un port derrière, afin de rediriger le trafic spécifique (adresse:port) vers une adresse ip local de notre réseau. En télé-travail uniquement, et via Teamviewer avec enregis...
besoin d'avoir un script php(perl/lua) pour créer un bridge en freeswitch pour initialiser un appel. si on appelle le script avec un numéro de téléphone alors on crée un appel entre une extension (3000) et le numéro appelé
...* Assister les utilisateurs bureautiques (Works, Word, Excel, PowerPoint, Publisher, etc.) * Mettre en œuvre une infrastructure de réseau, (TCP/IP, Hub, Switch, Routeur, Nat, Dmz, Firewall, VPN) * Diagnostiquer un dysfonctionnement, sécuriser un réseau et les données informatiques (NTbackup, Magic) * Traiter un appel et résoudre un probl&...
Bonjour, nous sommes une startup souhaitant mettre notre site web sur une infra Amazon. Nous cherchons un admin linux capable de monter cette structure, de mettre en place des NAT, de configurer un reverse proxy, et assurer les opérations de maintenance une fois le site en production (vérification de la capacité, a dispo, etc... ). Nous cherchons quelqu'un
Hi David ...need to update the router... have it configured to use one WAN connection for almost everything and the other one just for trunks from our server. ISP told us that it is about NAT set up but they don´t do it... The system is live so we may need to work on a tight schedule to set it, test it and if necessary restore if any fail appears...
IM using pptp, with nat, for my clients can access all the files and surfing internet trought vpn, but now the connection has made, but i cant do any this thinks!!, need to solved for used inclusive in android systems
...Now I'd like to develop a Voice Application using freewsitch for Voip Telephony. First I wan't to discuss what is the best architecture (Hardware, SO, Freeswich Version, freeswitch modules, failover, high avaliability), Install this Environment in my CloudServers, and then create a custom lua script that connects to my Artificial Inteligence API.
Hi! I love what you did with my fairy floss logo, thank you so much. I wasn't able to upload the photo of the back ground I wanted. So is there any way we could communic...way we could communicate about changing the background to a pastal coloured clouds background, and maybe even seeing the fairy floss stick itself rainbow coloured? Thank you, Nat
Hi i am looking for help on Big Blue Button Server setup, customization and branding along with performance optim...Blue Button Server setup, customization and branding along with performance optimization Knowledge of following Technologies may be required: Ubuntu, Web RTC, nginx, red5, FreeSWITCH, tomcat7, redis, Meteor, turn-stun coturn turn server
I need you to develop some software for me...based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid. Please include the past project info.
Hi there, I need some help with creating static NAT rules to allow my users to access our security camera software from wherever they are - not just on the internal lan.
...telecommunications and IT conusltancy business. Our business name is Asterswitch, coming from the two main platforms we offer support/consultancy with (i.e. ASTERisk and freeSWITCH) and it is a one man team. Our business provides help with PBX and network solutions. The main objective of the logo: - must be a simple, elegent, professional stamp - must
Mandatory Skills: 1) Hands on expertise in SIP, RTP and RTCP 2) Good knowledge on TUN,STUN, NAT 3) Free-switch working knowledge on audio conference using SIP and Web RTC 4) Good knowledge on RTP Proxy and routed audio conferences concept where media would flow via free switch RTP Proxy 5) Working experience of High Availability and Cluster 6) SDP
We need a standard single zone seven server cluster installation of Kazoo. Full setup of Kazoo with working modules
bbb-conf --check output is as below. And as you can see there are issues with WebRTC and HTM5 and for HTML5 I think also Mongod also has issues. If you have worked in Bigbluebutton and if you can help me fix this issue then provide your fixed price quote. BigBlueButton Server 2.0.0-RC4 (1518) Kernel version: 4.4.0-1060-aws
Hi, I need some assistance with AWS and the setup of routing between EC2 and Open VPN Server clients. I have attempted this but as soon as I turn off NAT in OpenVPN everything stops working. Setup I want is.. SQL Server EC2 Instance on a Private Subnet OpenVPN Access Server on a Public Subnet Clients Connected to OpenVPN Access Server (My Laptop for
I am looking for a freelancer to help me with my project. The skills required are freeswitch and fusionpbx FusionPBX CDR billing requirements Problem statement Currently i have a single FusionPBX configured in a multitenant environment with different SIP trunks from multiple providers configured per tenant. Given that there are multiple providers
...phone with Zoipher that registers to my SIP server over the WAN 2 x LAN connected SIP clients 1 x FreeSwitch Server on Windows You're free to change the setup, but this was working using FreeSwitch fine on my lan until i changed things over. Youll need: NAT & Voice skills on Cisco So far, I've been successful in getting all calls to work on the LAN
You need to configure a brand new fortigate 50e firewall. The device is connected already to our ISP and has a working internet connection in NAT mode. You need to configure the firewall to route in two different subnet, separated from each other. In the networks are NAS server cluster, VM's, Mail server, switches, EFPOS, FOG server etc. You
...system. We will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like Twilio, or FreeSwitch etc..If you Already have completed a Ringless voicemail drop system please bid. Please include the past project info. Thanks Skills required: Asterisk PBX, Telecommunications
...caller calls in. Our PBX platform is built on top of FreeSWITCH. We have a developer already that is very knowledgeable with FreeSWITCH and is going to help with this project. Here is a page that describes what we are trying to do. [login to view URL] Freeswitch does have an Event Socket Library that will allow integration
...CISCO ASA 5516X in active/active failover * setup outside VLAN with static IP * setup a 10.10.x.x inside VLAN * verify all wiring is correct. * setup anyconnect VPN * setup NAT for outside access for 2 servers * setup static IP for 2 servers and verify they have correct local IP and can access internet properly * verify all setup is valid correct. *
I've installed and got a running asterisk 13 from debian distribution. Added asterisk-gui and got that to sort of work but getting frustrated with what looks like sip/nat issue and that I couldn't find a way to be able to call between local extensions. It's mainly for personal use, I wanted a simple PBX that could handle a small group of sip devices
...manually added via apf -d are still being allowed through. Chkrootkit did not turn up any issues. service iptables restart produced the following messages Flushing firewall rules: [ OK ] Setting chains to policy ACCEPT: filter mangle nat [ OK ] Unloading iptables modules: [FAILED]
I'm using freepbx successfully with the Bria mobile app. However, this only works when connected t...to a ATT LTE connection, I don't receive any data on my Bria mobile app (so I cannot hear the other person on the line BUT the other person is hearing me). I guess this is a NAT/network setting on the free PBX server . I need your help to resolve this.
Server OS: Centos 7 Software: Asterisk 13 + FreePBX + a2billinng Network: Dynamic IP behind NAT Problem: - Call receiver no ring - No audio as it should to tell balance, dial 9 for sip or iax Because this desktop behind NAT, i'll provide Teamviewer or Anydesk access
Below are the key offerings as part of Contract. 1. Setup High available AWS infrastructure with best practices a. Public Subnets b. Private subnets c. NAT Gateway d. RDS e. Load balancer f. AutoScaling for Web tier g. S3 h. CDN i. AWS Inspector for Security vulnerability scanning j. AWS CloudTrail k. AWS GuardDuty 2. Deploy the application to AWS
WE NEED AN ASTERISK EXPERT TO INTEGRATE HYLFAX AS AN OUTGOING ONLY BROADCAST FAX SERVER. Fax Broadcasting is sending an identical fax to multiple recipients 1. SOF...MUST INCLUDE A NUMBER RECEIVERS CAN CALL AND DELETE THEIR NAME FROM OUR FAX LIST BY ENTERING THEIR FAX NUMBERS. MUST HAVE EXPERTISE IN: C, C++, AWK, VOIP, ASTERISK, PYTHON, FREESWITCH.
Freepbx general support, particularly security, external sip phone connections like csipsimple not working, obi202 not registering, vpn connection setup, etc It seems there is NAT / firewall config issue between remote extension and freepbx. Access available via TeamViewer only and credentials will not be shared for security and privacy purpose. No add-on
...0/24) and route all its traffic via VPN (incoming and outcoming) 4- Setup NAT port redirection for the VM to reach its services (SSH, FTP, HTTP) from the Internet Notes: 1- The hypervisor contain already another subnet of VMs ([login to view URL]) which need to work together with. 2- NAT port redirection need to be done via IPTables as it will be handled next
.../etc/ppp/pptpd-options ms-dns 188.8.131.52 ms-dns 184.108.40.206 ADD USER /etc/ppp/chap-secrets cbd pptpd 1234 [login to view URL] cbd2 pptpd 1234 [login to view URL] NAT iptables -t nat -A POSTROUTING -s [login to view URL] -o vir1 -j MASQUERADE iptables -t nat -A POSTROUTING -s [login to view URL] -o vir2 -j MASQUERADE /etc/[login to view URL] /sbin/iptables-restore < /etc/[log...
...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 (one-to-one and multi-user communications Users would
I'm looking for a tech who already has completed a Ringless Voicemail drop system. We ar...based company and will target users in US only so will calling 10-digit US phone numbers. We would ideally prefer some open source technologies to be used with it like FreeSwitch etc.. If you Already have completed a ringless voicemail drop system please bid
...that information is updated in a DB table. Adding -Look in the DB to see if that client ID already exists, if it does, prompt for new name -If not exist, enter client ID, NAT IP assigned, date/time and maybe some other details. Removing -Look for matching client/id in DB and remove it from the table. When the script ends, I I need to run one more