Someone pointed out that SIP won’t work as a webtransceiver replacement because SIP, unlike IAX, has no out of band signaling for PTT. I’m not sure how messaging works but I DO get a list of connected nodes on my CISCO SIP phone when connected to a node on my Asterisk/app_rpt PBX. -- Tim :wq On Aug 17, 2014, at 9:07 AM, Bryan Fields <Bryan@bryanfields.net> wrote:

On 8/17/14, 10:59 AM, mike@midnighteng.com wrote:

...

I would agree that a updated web-transceiver would be desirable, That html5 ver is also java based.

While java is simple and powerful, it also presents plenty of security problems.

You're wrong :)

It's Java Script and HTML5. Javascript has nothing to do with Java. It's confusing.

I just tried it out with the online demo. Worked perfectly in firefox 31 on OSX for me with the SIP number

http://sipml5.org/call.htm?svn=224#

Private ID : 1 Pubic ID: sip:1@sip.allstarlink.org Passwd: allstar Realm: sip.allstarlink.org.

Expert settings Disable Video X Enable RTCWeb Breaker X SIP outbound Proxy URL : udp://sip.allstarlink.org:5060 Disable Call Button Options X

Reload the phone page and register.

Put the node number (or 0) in the called number line and auth with your 10 digit pin.

Looks like we just need some "speed dial buttons" and a text message interface. I don't know how that messaging is implemented.

-- Bryan Fields

727-409-1194 - Voice 727-214-2508 - Fax http://bryanfields.net _______________________________________________ App_rpt-users mailing list App_rpt-users@ohnosec.org http://ohnosec.org/cgi-bin/mailman/listinfo/app_rpt-users

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