The first step to connecting our FreeSWITCH install to our newly provisioned Elastic SIP Trunk is to create a new external SIP profile in our FreeSWITCH configuration. FreeSWITCH is a highly featured platform with a large number of configuration files, the location of which will differ from platform to platform and from distro to distro. 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.
Cgm dve vol.1 for mac download. Take effective ways and guides to remove CGM DVE Vol.1+ on the Mac 1. Manually remove CGM DVE Vol.1+ on the Mac Option one: find the app's uninstaller to remove itself. Open Finder, select Applications in the sidebar; Search or find the app's uninstaller directly in the folder; Double-click on it on start the removal; If you cannot find the uninstaller, please take the following option to remove CGM DVE Vol.1+. CGM DVE Complete XXL is a set of 184 filters, transitions and generators for use in Final Cut Pro and Final Cut Express. In addition to the professional plug-ins the package also includes a series of Final Cut Pro workshops and 68 softwipe patterns (heart, star, clouds, bands, etc.).
ObSession = freeswitch.Session('sofia/192.168.0.4/1002')- Check to see if the call was answeredif obSession:ready then- Play file hereelse- This means the call was not answered. Check for the reasonlocal obCause = obSession:hangupCausefreeswitch.consoleLog('info', 'obSession:hangupCause = '. ObCause )if ( obCause 'USERBUSY' ) then - SIP 486- For BUSY you may reschedule the call for laterelseif ( obCause 'NOANSWER' ) then- Call them back in an hourelseif ( obCause 'ORIGINATORCANCEL' ) then - SIP 487- May need to check for network congestion or problemselse- Log these issuesendend. You can make the wav play when someone start a call, follow these steps. Place your wave into your freeswitch/conf folder.Add the code bellow to your freeswitch/conf/autoloadconfigs.Run a HTTP server that receives a POST request and returns your dialplan(which tells freeswitch to play your wav). Make sure your freeswitch/conf/autoloadconfigs/xmlcurl.conf.xml looks like thisHope this helps.
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. M-audio 2x2 driver. 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. 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.
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The first step to connecting our FreeSWITCH install to our newly provisioned Elastic SIP Trunk is to create a new external SIP profile in our FreeSWITCH configuration. FreeSWITCH is a highly featured platform with a large number of configuration files, the location of which will differ from platform to platform and from distro to distro. 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.
Cgm dve vol.1 for mac download. Take effective ways and guides to remove CGM DVE Vol.1+ on the Mac 1. Manually remove CGM DVE Vol.1+ on the Mac Option one: find the app's uninstaller to remove itself. Open Finder, select Applications in the sidebar; Search or find the app's uninstaller directly in the folder; Double-click on it on start the removal; If you cannot find the uninstaller, please take the following option to remove CGM DVE Vol.1+. CGM DVE Complete XXL is a set of 184 filters, transitions and generators for use in Final Cut Pro and Final Cut Express. In addition to the professional plug-ins the package also includes a series of Final Cut Pro workshops and 68 softwipe patterns (heart, star, clouds, bands, etc.).
ObSession = freeswitch.Session('sofia/192.168.0.4/1002')- Check to see if the call was answeredif obSession:ready then- Play file hereelse- This means the call was not answered. Check for the reasonlocal obCause = obSession:hangupCausefreeswitch.consoleLog('info', 'obSession:hangupCause = '. ObCause )if ( obCause 'USERBUSY' ) then - SIP 486- For BUSY you may reschedule the call for laterelseif ( obCause 'NOANSWER' ) then- Call them back in an hourelseif ( obCause 'ORIGINATORCANCEL' ) then - SIP 487- May need to check for network congestion or problemselse- Log these issuesendend. You can make the wav play when someone start a call, follow these steps. Place your wave into your freeswitch/conf folder.Add the code bellow to your freeswitch/conf/autoloadconfigs.Run a HTTP server that receives a POST request and returns your dialplan(which tells freeswitch to play your wav). Make sure your freeswitch/conf/autoloadconfigs/xmlcurl.conf.xml looks like thisHope this helps.
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. M-audio 2x2 driver. 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. 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.