Tag Archives: fax

Some recent projects for fun and work

I’ve had fun with a few personal and work-related VoIP projects lately. I’ll summarize them here, and if there’s any interest I’ll expand one or more of them into full how-to articles. 

speak2tweet
You may have heard about the Google/Twitter project called speak2tweet that was created over a weekend during the Egyptian protests, with the goal of allowing protesters whose Internet connectivity was cut to get their voices on to the web. Some folks on the PBX-in-a-Flash forum were talking about it and we realized it would not be hard to build, especially with the Tropo transcription service available to us.
Check out the first page of that thread for the dial-in numbers, and then try it out yourself. The code is posted on the forum for the first hack at it, which only does transcription. The current version includes links to the recordings, too, just like the original speak2tweet. I need to clean it up and post it. See and hear your spoken tweets at twitter.com/piafspeaktweet.
Fax
I have largely ignored fax over VoIP for a long time, because it’s not something I do. At PSU, we attach fax machines to analog lines; done. But recently, I was asked to assist with a quick-turnaround fax project: send a fax to 4,059 analog lines on campus, calls spaced one minute apart, and record the results, so that we can see how many fax machines are hooked up out there. Interesting. I tried Digium’s Free Fax for Asterisk module, but couldn’t get it to communicate. Next I built the soft-switch.org spandsp module, and told Asterisk to use it, and it worked right away. A Perl script using the excellent Asterisk::AMI module (referenced previously on this blog) drove the dialer, and standard Asterisk CDR with a few extra fields recorded the fax results. By the way, the faxes were sent in audio mode (not T.38) using SIP and G.711u over the Internet, and only 3 out of 4,059 calls failed.
FreeSWITCH revisited
Back when I wrote about using FreeSWITCH alongside Asterisk as a gateway to Google Voice, I determined that at some point I’d come back and dig into FreeSWITCH a little more just to understand it. I updated that article with a slightly better config, including lines that turn on comfort noise, which helps the FreeSWITCH-Asterisk bridge maintain synchronization on RTP. (See the FreeSWITCH wiki for more information on that.)