This can be anything, noting that it will be shown when you dial in ( screenshot). Once you’ve verified your account, the next step is ‘Create your domain name’.All of the input data can be fake except for the email address, as you need to click an account verification link ( screenshot). Fill in the account creation form with your email address, name, phone number, company name and size, and choose ‘Free SIP Services’ from the dropdown.At, click ‘Start Free’ in the top right corner of the homepage ( screenshot).They offer a lot of other services too, so signing up for an account can be confusing. Onsip offers a JavaScript-heavy browser-based SIP client that is free. Click the + under ‘Contacts’ on the left and add you also want a SIP account (to receive calls), you can use Linphone’s account setup form.Add the CCG SIP address as a contact so you can dial in with one-click.In Linphone it’s under Linphone > Preferences > Network > Transport – toggle “allow IPv6”.In Linphone it’s under Linphone > Preferences > Network > Network protocol and Ports.Update your settings to use a random SIP port (this will help avoid annoying calls).In Linphone it’s under Linphone > Preferences > SIP accounts > Default identity.Set your display name, which will be displayed in IRC.It should work out of the box, but there are optional configurations you can make. On Ubuntu/Linux, you can install it with sudo apt install linphone and launch it from the menu. Linphone is a SIP client that is free and available on many platforms.įor iOS, Android, Windows and Mac, you can download an installable file from the website. The following examples are SIP clients that members of the community have had success with. To do this, you need a SIP client installed on your device. We recommend you connect to our meetings with a SIP client. View the Project on GitHub w3c-ccg/ Connecting to a meetingįor the best experience, ensure you are connected through both audio and chat Audio Any help is appreciated.Increasing trust on the Web, one spec at a time. But the interface is still a bit cryptic to me, so I'm looking for some basic instructions to help me get acquainted with this world. And our phones appear to be mini-computers configured to look/work like a phone. I'm generally pretty tech savvy in the computer world. So it's not clear to me what we can and can't change and still have the phones work correctly with them.Īs you can tell, this is all a bit confusing to me. My understanding, though, is that when we set up OnSIP as our provisioning provider, they downloaded to our phones the software needed to work with their network. Any direction as to where to find that would be helpful. It would be nice to know which one we have. I was hoping for something a bit more aimed at those with little or no experience with this.īut for starters, there are repeated references to a phone's software (UCS) version. I'm afraid this reply still assumes a bit more knowledge about the phones and their programming than I am familiar with. I'm looking for enough detail to know what menu on what device I need to go into, and then where to find such parameters so I can change them. I asked OnSIP about it, but they claim it's a Polycom issue, not an OnSIP issue. So I just need to know how to enable this feature. I figured out how to connect to the phone via the Web Configuration Utility, but after looking through all the settings there, I can't find it there either. I've looked through several user and admin guides, but they all seem to assume that you already know how to do this. The problem is, I can't figure out how to enable or change this parameter. After doing searches on this, there are articles on using a feature called "up.OffHookLineView.enabled" to make the Lines view the default when off hook. We have to choose Lines to see the users instead. When transfering calls, instead of seeing the list of lines in the office, we first see a list of SIP addresses. We use VVX 410 and 501 phones with OnSIP. I've searched through FAQs, but can't find the answer to my specific question. Brand new VoIP user, so pardon my ignorance here.
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