To see if any process is currently bound to port 80. If so, check to see if another web server is installed. If so, then stop the web server and try to restart nginx. The main purpose of the demonstration configuration in FreeSWITCH is to showcase all. Make a test call, and access all of the functionalities of FreeSWITCH.

I work for an ITSP (VoicePulse), and much like the rest of the VoIP world, we've used Asterisk more than FreeSWITCH, but collectively, we have experience with both. I haven't really paid much attention to FreeSWITCH in the last several years, so keep in mind that it's possible some of my opinions might be out of date.That said, is right in saying that they're fairly comparable in terms of features.

I think you'd be hard pressed to find anything that you could do with one which you couldn't accomplish with the other. Asterisk tends to be lighter on hardware requirements, and has a much larger support community. FreeSWITCH on the other hand does allow for higher call capacity given identical hardware (or at least it used to, I haven't run a comparison of recent versions), and makes things like multi-tenancy significantly easier. FreeSWITCH also uses XML for the configs, which can be kind of annoying if you find yourself making frequent changes.I started out using Asterisk because that used to be the only option back in 2003, and it was (and still is) great software that does its job very well. We built some FreeSWITCH boxes about 5 or so years ago just to try it out, and they're still up and handling a decent amount of calls today with very little required attention. Ultimately we primarily use Asterisk though, because while FreeSWITCH functions perfectly fine, I didn't really see a huge advantage to using it over Asterisk and in the end we're just more familiar and comfortable with Asterisk.You didn't really say anything about your application, so I can't make much of a recommendation based on that, but I'd say if you are in the VoIP industry and will be dealing with PBXes frequently, it might be worth having some experience with FreeSWITCH. However if this is something that you're just want to get setup, working, and forget about it, I'd say go with Asterisk to leverage the larger community as it will make your life easier in the long run.

There are web UI's you can overlay on either one, so that isn't much of a factor either. 13.7.0-rc1 was released yesterday, so it might be a week. We usually like to give the RC process a bit of time.BLF is a lot more than just RFC 4235, which - by the way - is supported via respjsipdialoginfobodygenerator (for the dialog-info+xml event packages). And it works just fine on my phones, at any rate:-). Do you have a specific ASTERISK issue you're referring to?The Opus codec is a fair point, since Asterisk doesn't include it.

Of course, we haven't claimed that we do support Opus - as pointed out, that's something FreeSWITCH has.As far as locking up/memory leaks go, I'd be curious which ones that were fixed recently or in the bug tracker that you've personally run into. Sure, we've fixed major issues in some recent releases, but pointing out that fixing bugs is evidence of instability makes it very hard to ever be stable. Last time I checked, most software fixes bugs in their releases. And if you look at the bug tracker, I'm sure you'll find more that we haven't fixed. It's an open source project, that's kind of how things go.But as I said at AstriCon, we're using Asterisk 13 in production. Take that for what it's worth, I guess. Well, that is fixed in 13.6.0 at least:-)The unfortunate thing is that to really get the URI size that you'd like in Asterisk, you have to configure PJSIP to accept a larger URI size as well.

There's a few other things that I like to configure in PJSIP as well - larger allowed packet sizes, IPv6, etc.I think it kind of sucks that you have to tweak PJSIP at all. Ideally, Asterisk users would be completely insulated from configuration in PJSIP. Unfortunately, that's the result of going with 'thou shalt not embed PJSIP' - which was a long and contentious debate on the. I can't say I disagree with the result that we ended up with, but there's just hard repercussions that we're still working through.I'm hopeful that we'll get through some of that in the next year, in some way or another. Not in the least.

The chansip driver is a bloody mess to put it mildy. It gets the job done - most of the time - as long as you don't need to change the code.

But it doesn't do sip 'right' - it does it the way a pbx engine that wants to be in control all the audio streams would do it. It has honestly been holding Asterisk back from doing things right.

Catalina album artwork. So yes, it's been a case of chanpjsip allowing Asterisk to catch up to other implementations. Even with PJSIP though, Asterisk is not a sip router - for that you should use something else like Kamailio.

Session: answer; - answer the call - register the sound input callbacksession: setInputCallback( function onInput( s, type, obj) end);- start to detect the speech to keep firing the feed interface insite yyasr modulesession: execute( 'detectspeech ', 'yyasr directory directory ');- resume the detectspeechsession: execute( 'detectspeech ', 'resume ');- stop the detectingsession: execute( 'detectspeech ', 'stop ');- check./script/yyasrdemo for details usage. Secondary development TTS module. TTS module structure.