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thirdhatch wrote:We figured out the issues with PBXWebPhone. But we are having trouble getting it to work with Firefox. It only works with Chrome.
ngtechnologies wrote:thirdhatch wrote:We figured out the issues with PBXWebPhone. But we are having trouble getting it to work with Firefox. It only works with Chrome.
We upgraded to PHP7 in our web server (actually tried to upgrade PHP5 to PHP7 and broke Vicidial, was new install with PHP7 installed before any Vicidial or prerequisites were added) and works Chrome and Firefox, but not IE or Edge. So far we have had 70 agents logged in and working with no issues.
Scratch install Vicidial on Centos6
1 database server
1 webserver PHP7
4 asterisk boxes
1 admin server/database slave
chornyi_taras wrote:ngtechnologies wrote:thirdhatch wrote:We figured out the issues with PBXWebPhone. But we are having trouble getting it to work with Firefox. It only works with Chrome.
We upgraded to PHP7 in our web server (actually tried to upgrade PHP5 to PHP7 and broke Vicidial, was new install with PHP7 installed before any Vicidial or prerequisites were added) and works Chrome and Firefox, but not IE or Edge. So far we have had 70 agents logged in and working with no issues.
Scratch install Vicidial on Centos6
1 database server
1 webserver PHP7
4 asterisk boxes
1 admin server/database slave
so you have patched asterisk to have webphone working in Firefox?
regarding Edge support- sipJs supposed to support it in nearest future https://github.com/onsip/SIP.js/issues/432
https://sites.google.com/a/chromium.org/dev/Home/chromium-security/deprecating-permissions-in-cross-origin-iframes#TOC-To-continue-to-use-permissions-from-iframes-on-your-website...
mflorell wrote:Asterisk 13 is going to be required with new versions of Chrome if you want to use WebRTC.
In addition to this, I also added the microphone permissions flag to the webphone IFRAME in the vicidial.php svn/trunk codebase, so if you are getting the permissions error, you will want to upgrade to a more recent svn/trunk revision.
mflorell wrote:How did you install Asterisk 13?
We do have a package within OpenSuSE, but here are the last working from-source instructions I used:
http://vicidial.org/docs/ASTERISK_13.txt
mflorell wrote:Just send in an email to support@vicidial.com to get the draft instructions for how to use the beta vicidial webrtc phone.
Since WebRTC is not yet a finalized standard, Google has decided to muck around with requirements for it, so soon in Google Chrome you won't be able to use a WebRTC phone with an Asterisk 11 server, due to changes in "requirements" which are not supported in Asterisk 11. So, you will need to use Asterisk 13 instead. We have clients running it now in production, but these recent changes have thrown a wrench into our official release plans until we make a few changes to our code and the instructions on the Asterisk side.
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