all,Trying to do my first WebRTC. Using stock asterisk 1.13.0. I setup the asterisk according to the recipe on the wiki, but cannot get it to work. Dialing from sipml5 on chrome I get no sound, regular bria on standard sip works.My network setup by ..
Author : Antonio Gómez Soto
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I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, Igot nothing.My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital oce..
does anyone have a recommendation for a SIP phone, which allows dialing from a phonebook, and hiding the dialed number from the end users? Also from the call history of course.It seems Mitel can do this, and I have a use case where this is a requirement.Than..
I am trying to setup a WebRTC connection to asterisk 1.13.0. Using Bria a regular SIP connection works, but using sipml5 on chrome, Igot nothing.My network setup by the way: I am working behind a comcast cable modem, the test setup is at digital oce..
The asterisk wiki page says:Sorcery.conf allows you to try to configure other PJSIP objects such as transport using realtime and it currently wont stop you from doing so. However, some of these object types should not be used with realtime and this ..
I am slightly confused by the difference between chan_sip and pjsip. Especially the new (to me) objects aor and contact.I am having trouble mapping them to the typical SIP configuration settings on a phone.Suppose I have a phone with two line butto..