FreeSWITCH package for pfSense 1.2.1 and 2.0 released. PBX or Proxy
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@mcrane:
The package does add files to /usr/local/lib but does not remove them on when the package is removed because I don't know if some other package is dependent on them.
BTW: I reinstalled FreeSWITCH and imspector starts again.
I guess that if I uninstall only FreeSWITCH now imspector shouldn't stop working if FreeSWITCH doesn't remove anything. I'll try that later and if imspector still stops working I'll report back.
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And to get SIP server listening on both interfaces I have copied the whole Internal profile tab and pasted it over to External profile tab with these changes:
Internal:
<param name="rtp-ip" value="192.168.0.84">
<param name="sip-ip" value="192.168.0.84">
(192.168.0.84 is my LAN IP on pfsense)
External:
<profile name="external">The problem:
Communication of LAN-LAN users works great, WAN-WAN as well, calling from LAN to a user who is connected to WAN works as well, but (!!) when a WAN-connected user wants to call a user connected to LAN he gots "the person at extension xxxx is not available"</profile>Internal profile is the profile that handles registrations. Most often used for extensions but can also be used to connect to by other devices such as FreeSWITCH.
External profile does not do registration.
Their names are a little misleading. By default FreeSWITCH binds to the WAN on pfSense for both Internal and External profiles. If you don't have a static IP address then you would need to use dynamic DNS with pfSense so that the location of the phone system would be constant. Phones on LAN can by default reach the WAN ip where FreeSWITCH is bound to without problem. Phones from the Internet need ports opened for the WAN interface but don't require NAT since its on the WAN.
Earlier this week others have requested ability to connect when using VPNs. To do that FreeSWITCH needs to be listening on an internal address that is reachable from the VPN. This will require copying a profile similar to what you have done. I don't suggest changing the External profile instead copy the internal profile and make a new one from it called something like 'LAN'. Inside the profile you would also need to change the name and remove the alias so that it doesn't conflict with the internal profile. Then restart freeSWITCH.
Soon as I have time I will be creating a new tab called 'Profiles' and will move the Internal and External tabs to that tab. Then will make it possible to manage profiles from there. I'm also thinking add a default to the new installs that include the LAN profile by default. Then people would be able to register to the LAN IP which is what most people are likely used to.
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I for one would definitely be for binding to the LAN interface by default.
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@mcrane:
And to get SIP server listening on both interfaces I have copied the whole Internal profile tab and pasted it over to External profile tab with these changes:
Internal:
<param name="rtp-ip" value="192.168.0.84">
<param name="sip-ip" value="192.168.0.84">
(192.168.0.84 is my LAN IP on pfsense)
External:
<profile name="external">The problem:
Communication of LAN-LAN users works great, WAN-WAN as well, calling from LAN to a user who is connected to WAN works as well, but (!!) when a WAN-connected user wants to call a user connected to LAN he gots "the person at extension xxxx is not available"</profile>Internal profile is the profile that handles registrations. Most often used for extensions but can also be used to connect to by other devices such as FreeSWITCH.
…yes, I know that, I just used the profile filename so I could edit it using the web interface. Just imagine I have the origin profile internal and it's copy running on another ip address, both are shown in status and running ok. The only problem is routing between these profiles when someone from WAN calls to LAN. Calling from LAN to WAN works.
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What kind of phones are you using on the remote side?
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@mcrane:
What kind of phones are you using on the remote side?
x-lite
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Well, after 8 months of working waayyyy too many hours, I finally have some extra time to play with FS, so I upgraded, and ran right smack into my same original problems with caller ID. After fighting with it for way too long, I bring you a solution. If you're using Voicepulse, make sure your outbound_caller_id and outbound_caller_name match, and are 10 digit numbers. Simple, huh? I knew this 8 months ago, just forgot. … :P
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Fun scenario…
Call to PSTN number succeeds from phone connected to Linksys PAP2.
Make call to extension on default extension from phone connected to Linksys PAP2.
Hang up while call still ringing
Attempt call to same PSTN number
Call will failWhy?
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Fun scenario…
Call to PSTN number succeeds from phone connected to Linksys PAP2.
Make call to extension on default extension from phone connected to Linksys PAP2.
Hang up while call still ringing
Attempt call to same PSTN number
Call will failWhy?
I don't think you have provided enough information to answer this but I can say that it works fine with my Linksys PAP2T.
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pfSense FreeSWITCH package version 0.8.7.4 is now available.
Latest improvements include:
SIP profile management.
Default LAN SIP profile that binds to the LAN IP address. This enables registration to the LAN IP.
Textareas now use editarea which provides syntax highlighting, line numbering and more.
Dialplan default.xml, public.xml and vars.xml textareas now have 'Restore Default'
Status page now shows all SIP profiles.
tail command for log viewing on the 'Status' tab increased to 500 lines, also using editarea
and more… -
Hey mcrane,
Just wanted to let you know, I was playing around in freeswitch trying to setup inbound faxing, and ran into an issue with mod_fax not loading properly. I was able to get things sorted eventually by trying to load the mod_fax.so object manually from fs_cli.so; when I tried that it complained that /usr/local/lib/libspandsp.so.2 didn't exist, and indeed it doesn't, at least not on 1.2.3-RC1. Symlinking libspandsp.so.1 to libspandsp.so.2 appears to have fixed everything right up for me though, although I'm not sure that's the "right" way to get things working.
Thanks again for all your work on the pfsense package.
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Just wanted to let you know, I was playing around in freeswitch trying to setup inbound faxing, and ran into an issue with mod_fax not loading properly. I was able to get things sorted eventually by trying to load the mod_fax.so object manually from fs_cli.so; when I tried that it complained that /usr/local/lib/libspandsp.so.2 didn't exist, and indeed it doesn't, at least not on 1.2.3-RC1. Symlinking libspandsp.so.1 to libspandsp.so.2 appears to have fixed everything right up for me though, although I'm not sure that's the "right" way to get things working.
Thanks to your post I was able to load mod_fax. I didn't stop there I immediately started to build a GUI interface for it. Should be done sometime this week. So lists the fax as a tiff file for download, and converts the fax to png and pdf. FAX to email and a method to send the fax is next.
Best Regards,
mcrane -
Can I request one feature? A quick button that resets all settings to default.
I had everything working for the last while (still have not had time to play with IVR), and I did a freeswitch upgrade and now my incoming calls go right through to voicemail. I can't even do local transfers.
I am guessing the reason is that I did some changes to the original settings to force freeswitch to my Lan ip's to get registrations working. Due to all the changes done to freeswich (looks good, interface is much better), I see that my changes could be the problem. Outbound calls to provider work, incoming go right to voicemail now (not available), and internal calls are not available. It was working previously.
Honestly the easiest way would be to start with default settings and set things up again. If I uninstall and reinstall, all the settings are saved. I tried modifying the settings file, and always seem to corrupt it.
I can't see that I am the only one who may want to hit a button to reset to mcrane default settings.Update. Ok, I just need to look some more. Looks like mcrane has already done this. Wow, snuck this in on me… :)
Update 2. Defaults did not make things better. Still getting errors on internal transfers in the log..
IP 192.168.0.6 Rejected by acl "domains". Falling back to Digest auth
So much for things running smoothly for the last couple of months..Update 3. Restore settings on the Lan.xlm does not work.
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Can someone PM me with a cut/paste of their lan.xml?
My test machine is busy right now, and I can't delete the drive and install a fresh pfsense/freeswitch right now.
I'm hoping that will restore my internal and inbound calling…Thanks!
Update. The restore does work on the lan file. The confusing part was that it put my ip in the file, so I thought it was my modified one (I had to put the local ip in the internal file and then I could get internal calling).
ANyways, I'm back to still not working. Whatever the changes were to the last couple of releases were, killed my internal and inbound calling. Not sure why internal callings would fail, and send it to voicemail.
Is it normal to have no internal registrations, and all the phones on the Lan? (i would assume so)
Also the log is getting user not registered on to of the rejected by acl domains even though the phones are clearly registered under the lan.I'd install an older version if I could as things were working before.
Oh well, I guess this is the fun of working with these packages!
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tester_02: a tool that would help see what is going on is to use the freeswitch console.
From the pfsense console do the following:
cd /usr/local/freeswitch/bin
./fs_cli -H 192.168.1For more help please use IRC -> freenode -> #pfsense-freeswitch
IRC clients that work well
Firefox add on chatzilla
xchat
http://www.mibbit.com/chat/ (online IRC client) -
tester_02, I've had the same symptoms when registered on LAN profile. I have just used the WAN address to register my ATA and all works well registered to the Internal profile.
db
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Introducing pfSense FreeSWITCH package version 0.9.
Whats in the new release.
1. FAX
FAX interface it includes fax to email consider. The FAX feature is currently beta until more people test it and report back. I have done my some testing and found that some VOIP providers do better with faxing than others. Viatalk seems to do pretty good. I suggest people help out testing this and report back how it performs with their provider.2. Hunt group
a. fix warning messages
b. fix ability to delete from the hunt group list
c. add sip uri examples.3. Auto Attendant (IVR) add the following fields
a. Call timeout is the time allotted to ring the destination before going to voicemail (for extensions).
b. Direct Dial can be enabled or disabled. If disabled then only the auto attendant (IVR) options numbers will be allowed from the auto attendant. If enabled extensions and feature codes not defined in the IVR can be called.
c. Ring Back allows you to choose whether to provide the caller with ring tone or music on hold.
d. Add examples to sip uri.
e. Add edit area tool to 'Javascript Condition' which adds javascript syntax highlighting, a full screen option, and more. -
Installed and working mostly okay! Can't wait to play with the fax but that will have to wait for another day. Glad to see the other things added - all very welcome additions, the ring versus music on hold is nice. Hunt group is huge, that's something I'll be playing with right away as well.
One issue I have been having - and had with asterisk but to a lower extent:
2 phones both registered to my pfsense freeswitch. (no voip provider in this example)
I call between phones - after 20 minutes, there is a 3 second delay in the conversation. Makes the call terrible even though the quality is fine. I assume it will continue to get worse if I let the call go on. I drop the call, and start it again and everything is fine.I can duplicate this with a call from an external (5km) voip phone to an internal phone, and also from a external voip provider to an internal voip phone. Duplicated on different internal devices and two external providers.
pfSense box is a AMD Geode that doesn't seem to be very loaded CPU wise to me. Using G711 codecs. Haven't really played with changing that.
Does anyone have any initial thoughts? pfSense is running snort and squid. Traffic shaper is configured with the wizard to give voip priority. Using a fairly new 1.2.3.
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I've noticed the issue you mentioned. I was in a 1.5 to 2 hour long conference call muted until the end of the call at the end of the call there was a 5 - 7 second audio delay on one side.
I just started using Jitterbuffer to see if it helps.
http://wiki.freeswitch.org/wiki/JitterbufferMore information
http://www.freeswitch.org/node/57 -
FreeSWITCH package 0.9.1
a. FreeSWITCH package add 'd' for default option to IVR which is the option that is performed if the caller dtmf doesn't match any other option (Direct Dial must be disabled). Other options that have already existed 't' timeout which determines where to send the caller if no dtmf is detected. 'n' for now this is useful for the during office hours if you want to direct the calls to ring an extension or huntgroup immediately and not wait for any dtmf keys to be pressed.
b. Added a new profile for hunt group and auto attendant destination extensions. The 'auto' profile searches all profiles to find the one the user is registered to and then directs the call to the profile that it found.
c. Extensive testing and fixed a few minor issues bugs with the IVR.