Hi everyone, I'm looking for someone who is familiar with C++ and with SIP to help me with updating this software and make some necessary changes. I need 3 changes made to it. 1) It's very outdated and needs to be updated with all the current packages and frameworks 2) Right now when an incoming calls comes in Outcall opens up a window in Outlook but
Necesito una app para Android. Me gustaría que la diseñen y la creen.
We have RedHat Linux release 6.5 (Santiago). We had an old version of curl installed to do API calls but it doesnt have the right TLS protocol so we need to upgrade it to v7.60.00 We have installed this but then we get errors saying https is disabled in libcurl. I tried to reinstall with ./configure --with-ssl=/usr/local/openssl (which I also installed)
This is a repost! I need an application that 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. The app should be able to run on cheap devices (+-100$). I was thinking on a way to emulate a headset
i am working on an app that can provide unlimited calling from Canada to India( same functionality as Rebtel). I looking for someone who can provide pstn gat...functionality as Rebtel). I looking for someone who can provide pstn gateway solution in Canada with unlimited incoming. Any type of pstn gateway solution can be considered( ex. pri, sip......)
Looking for someone that has SIP/PJSip related experience to take our current PJSip plugin that works in NAT64 environments to work in a IPv4 only environment. The current code base is using this patch to work in NAT64/DNS64 environment: [login to view URL] This code works when on an IPv4 and IPv6 network, or IPv6 only network,
I need to configure SIP server which supports API's for Inbound and Outbound also i need live call data based on DID
...are building a freepbx server in Canada that can receive inward and outward calls and need some advising setting up a pstn gateway. For example pri line would be better or a sip trunk or if there is any other option ? Looking for a knowledgeable person who can get this project started. p.s. we will be posting some more jobs for the same so will need
Www.optionsnoob.com. The warning message that neds to be resolved: You currently have TLSv1 enabled. This version of TLS is being phased out. This warning won't break your padlock, however if you run an eCommerce site, PCI requirements state that TLSv1 must be disabled by June 30, 2018.
...NET::ERR_CERT_AUTHORITY_INVALID error. My virtual host looks like following: <VirtualHost *:443> SSLEngine On SSLCertificateFile /etc/pki/tls/certs/[login to view URL] SSLCertificateKeyFile /etc/pki/tls/private/[login to view URL] ServerName [login to view URL] ServerAlias [login to view URL] DocumentRoot /var/www/html/example <IfModule mpm_...
...of my webcams. I Need a DLL Class that I can use as reference in my C# Desktop Application with the possibility to : "Make or Receive a Video Conference Call using H323 or SIP" In this class I would set properties like: VideoDevice (Select a webcam) AudioDevice (Select a microphone/line in) Volume Bandwidth Video BitRate Bandwidth Audio BitRate
...custom Outlook addin to connect to a SIP extension and when an incoming calls comes in it should give a pop up notification with the contacts info from outlook. It should be exactly like this: [login to view URL] but it should be able to connect to any asterisk SIP server. You can even use this addin
...the following line of code in the apache configuration file: SSLProtocol -all +TLSv1.2 This will disable all protocol except for TLS1.2. However, it you still want the other TLS versions enabled for backwards compatibility, the following line should work: SSLProtocol -all +TLSv1.2 Alternatively, if you're using Nginx, the lines would be similar to the