FreeSWITCH package for pfSense 1.2.1 and 2.0 released. PBX or Proxy

cybrsrfr

I have just released a FreeSWITCH package for pfSense 1.2.1 also works on pfSense 2.0. It requires pfSense 1.2.1 or higher because it was compiled for FreeBSD 7.

What FreeSWITCH can do for you:

SIP Proxy
Soft-Switch
B2BUA
PBX with:
IVR (auto attendant)
Voice Mail
DISA (Direct inware system access)
Conference Server
Text to Speech
Speech Recognition
Auto Dialer
Audio can be 8khz standard, wideband 16khz, or ultrawideband 32 khz
SRTP Support
TLS support

And best of all its customizable

New Screen Shots are available:
http://portableusbapps.com/images/FreeSWITCH/

General Info on FreeSWITCH
http://www.freeswitch.org/
http://wiki.freeswitch.org/wiki/Main_Page

SIP Proxy

http://wiki.freeswitch.org/wiki/Proxy_Media

PBX

To get started setting this up as a PBX setup a few extensions the default config accepts extension numbers between 1000 and 1019. The extension numbers can be changed are noted in the package GUI on how to make the change. Once you have setup an extension you can call Music on Hold by dialing 9999.

5000 Sample IVR
9996 for an echo test
9995 for a 5 second delay echo

Setup another extension with a phone and you can then call the other extension by its number.

To call out to public telephone numbers you will need a FXO or a VOIP provider. To configure a VOIP provider use the Gateway tab. The field names have been matched up to the xml tag names. Under Gateways (aka Providers) the field names match exactly with the xml tag names shown at: http://wiki.freeswitch.org/wiki/SIP_Provider_Examples this helps keep the documentation and examples on the wiki relevant to the pfSense FreeSWITCH package. The GUI has been built to increase your knowledge of the underlying XML config files.

Then to use the Gateway for an outbound call you need to direct the outgoing call to the gateway through the 'Dialplan' tab.
Here is an example for a gateway named asterlink.com you will want to put it near the bottom below hold_music and just above the warning message. Also remember to change asterlink.com to the name of your gateway (provider). Also replace the area code from my area code of 208 to match yours that will enable 7 digit local North American dialing.

^(\d{7})$ is a regular expression representing any 7 digit number.

To receive inbound calls you need to direct inbound numbers to your gateway you can do this by editing the config from the 'Public' tab. The default example is fairly intuitive I will expand on it a little first a term DID (direct inward dial) this is essentially the phone number the provider has given you.

The ^(5551212)$ is a regular expression representing the inbound phone number often this is a 10 digit number.
<condition field="destination_number" expression="^(5551212)$">The transfer application example below is set to transfer the incoming call to extension 1000. This could also be an extension of given to an IVR under the 'Dialplan' tab.
<action application="transfer" data="1000 XML default">Additional Info
http://wiki.freeswitch.org/wiki/Getting_Started_Guide

That should be enough to get started and the wiki can take you even farther.
http://wiki.freeswitch.org/

Enjoy!</action></condition>

jigpe

Thanks master! Im exploring it now :)
You`re the man! :)

I`ll inform you the result of my explorations :)
Good luck!

jigpe

greyman

I have installed freeswitch to look through its configuration on my pfsense installation (built on Wed Oct 8 17:46:25 EDT 2008 FreeBSD 7.0-RELEASE-p5). Once installed I noticed that it show up under services and can be controled but under packages is does not show up as installed or something to be installed.

cybrsrfr

You should see FreeSWITCH in the following locations:

//before installation
System -> Packages -> Available PFSense 1.2.1-RC1 packages

//after installation
System -> Packages -> Installed Packages
Services -> FreeSWITCH
Status -> Services

On my test systems it does show up under System -> Packages -> Installed Packages.

If for some reason you can't see it under: System -> Packages -> Installed Packages
You could try installing over top of it using:
http://x.x.x.x/pkg_mgr_install.php?id=FreeSWITCH
Change the x.x.x.x to the IP address of your pfSense firewall.
Make sure not to interrupt the package installation.
Then check again to see if is shows under 'Installed Packages'.

thekod

Great package so far…fighting with it a bit though. Doesn't seem like it reloads the config when you save changes. Is that desired behavior?

cybrsrfr

It runs FreeSWITCH's reloadxml on all changes which is good for re-reading the config. However for a new Gateway to be applied I have usually restarted FreeSWITCH. I've just asked on FreeSWITCH forum and they said a reloadxml then restart the sofia sip profile would work. I will get that changed soon.

thekod

Sounds good. I confirmed that the reloadxml command is working, although it didn't work for me when adding extensions immediately after install, but that might just be FreeSwitch trying to be efficient and not having Sofia loaded (since I didn't have any sip devices listed prior to that), or who knows. It's working now, so whatever.

Another question…you'll be getting several package bug reports from me (if there are any bugs and I'm not just being retarded. :) ), do you prefer I PM them to you?

#1: When changing the area code in the settings tab, 7 digit calls still get prepended with 918, due to the <x-pre-process cmd="set" data="default_areacode=918">line in the VAR tab. Should this be changed to the value from the Settings tab? EDIT: I actually put ${default_areacode} in the area code field on the settings page...I figure that should pull it from the vars.xml file, where I have it set correctly. Is this correct? Also, what xml setting does the "area code" on the settings page change?

#2: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml, 10 digit calls just drop. Adding the following section will fix that by prepending a 1, and sending the call out.

  <extension name="domestic.provider.com"><condition field="${toll_allow}" expression="domestic"><condition field="destination_number" expression="^(\d{10})$"><action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"><action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"><action application="bridge" data="sofia/gateway/${default_gateway}/1$1"></action></action></action></condition></condition></extension>

#3: I can't get the "say:" command in the IVR xml file to work. I get the following error in the console:```
2008-10-28 22:43:15 [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:Press 1 to join the conference, Press 2 to join the other conference.PCMU


#4: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml all the caller id setup is done with single dollar signs…IE, ${outbound_caller_id_number}.  I'm no FreeSwitch guru, so I can't explain why, but I can't pass callerid through Voicepulse to the PSTN until I change them all to double dollar signs, as I saw in the vars.xml file...note that I didn't add a dollar sign to ${default_gateway}, only the caller_id settings, and that's apparently working...I'm not sure what the difference is.

#5: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml all the caller id variables reference ${outbound_caller_id_number} or =${outbound_caller_id_name}.  In the vars.xml file, the only caller id variables are outbound_caller_id and outbound_caller_name.  Until I assign my caller id data to the first variables, outbound_caller_id_number and outbound_caller_id_name, I can't pass callerid through Voicepulse to the PSTN.</x-pre-process>

cybrsrfr

I have completed a few more changes to the package.

1. When adding a Gateway the following command is now called automatically through a socket connection.
sofia profile external rescan reloadxml
This will pick up new gateways that have been added or deleted and handle them appropriately without restarting FreeSWITCH. There still may be times a few times that require a restart for a gateway. I could have been more aggressive and had it run: sofia profile external restart reloadxml but that would interrupt current incoming/outgoing calls so I chose not to go that way.

2. Added a few more input fields for Gateways such as 'Realm' and 2 others fields in many cases this is not needed but I wanted to make sure that there is flexibility in GUI for special situations. If a field is left blank then the corresponding xml parameter tag will not be used in the config. The fields under the 'Gateways' tab match up exactly with the SIP provider examples at:
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples

3. The 'Modules' tab has been changed from the original text area to a list in the GUI that allows you to enable or disable different modules. Note: not all modules are available in this build… an example would be Cepstral text to speech which doesn't have a native FreeBSD build. Flite text to speech is available if its modules is enabled.

Now to answer your questions...

@thekod:

Sounds good. I confirmed that the reloadxml command is working, although it didn't work for me when adding extensions immediately after install, but that might just be FreeSwitch trying to be efficient and not having Sofia loaded (since I didn't have any sip devices listed prior to that), or who knows. It's working now, so whatever.

Sofia (SIP) is loaded by default. If you restart FreeSWITCH under Services then you should check the Status tab to make sure that the sofia internal profile is loaded if it does not load it will say 'Invalid Profile!' This is caused from a port not closing by the time you restart FreeSWITCH. It can be solved by shutting down the service longer sometimes up to 2-5 minutes then starting it again. Rebooting will also work to clear that up.

@thekod:

Another question…you'll be getting several package bug reports from me (if there are any bugs and I'm not just being retarded. :) ), do you prefer I PM them to you?

If the information can be valuable to others then the forum. Items that are not likely to be useful to others you can pm me.

@thekod:

#1: When changing the area code in the settings tab, 7 digit calls still get prepended with 918, due to the <x-pre-process cmd="set" data="default_areacode=918">line in the VAR tab. Should this be changed to the value from the Settings tab? EDIT: I actually put ${default_areacode} in the area code field on the settings page…I figure that should pull it from the vars.xml file, where I have it set correctly. Is this correct? Also, what xml setting does the "area code" on the settings page change?</x-pre-process>

$${default_areacode} in the area code field on the settings page should work fine.
The settings page populates the area code in the following xml file /usr/local/freeswitch/conf/directory/default/default.xml

@thekod:

#2: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml, 10 digit calls just drop. Adding the following section will fix that by prepending a 1, and sending the call out.

  <extension name="domestic.provider.com"><condition field="${toll_allow}" expression="domestic"><condition field="destination_number" expression="^(\d{10})$"><action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"><action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"><action application="bridge" data="sofia/gateway/${default_gateway}/1$1"></action></action></action></condition></condition></extension>

This can also be done from the dialplan tab as noted in the this first message in this thread. As you have shown the caller id can also be done for all calls going outbound to the gateway with the 10 digits shown above. The other method for caller id is per extension which can be done by setting the effective caller id on the 'Extension' tab.

@thekod:

#3: I can't get the "say:" command in the IVR xml file to work. I get the following error in the console:```
2008-10-28 22:43:15 [ERR] mod_native_file.c:68 native_file_file_open() Error opening /usr/local/freeswitch/sounds/en/us/callie/say:Press 1 to join the conference, Press 2 to join the other conference.PCMU

I have not worked with the XML IVR because I prefer doing the IVR in javascript. For help with the XML IVR see the wiki at http://wiki.freeswitch.org/

To see an example IVR in javascript.
http://wiki.freeswitch.org/wiki/Javascript_Examples look for the example ivrmenuofficehours.js link.

The sounds under /usr/local/freeswitch/sounds/en/us/callie/ are pre-recorded messages using Cepstral's text to speech engine.
At this time Cepstral does not have a native FreeBSD build. Alternatives are to pre-record messages on a different computer, use Flight text to speech engine, or make your own voice recordings.

@thekod:

#4: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml all the caller id setup is done with single dollar signs…IE, ${outbound_caller_id_number}. I'm no FreeSwitch guru, so I can't explain why, but I can't pass callerid through Voicepulse to the PSTN until I change them all to double dollar signs, as I saw in the vars.xml file...note that I didn't add a dollar sign to ${default_gateway}, only the caller_id settings, and that's apparently working...I'm not sure what the difference is.

Looks like you found a missing $ in the default config. Next time I compile I will check to see if that has been corrected if not I will report it if you don't beat me to it.

@thekod:

#5: In /usr/local/freeswitch/conf/dialplan/default/01_provider.com.xml all the caller id variables reference $${outbound_caller_id_number} or =$${outbound_caller_id_name}. In the vars.xml file, the only caller id variables are outbound_caller_id and outbound_caller_name. Until I assign my caller id data to the first variables, outbound_caller_id_number and outbound_caller_id_name, I can't pass callerid through Voicepulse to the PSTN.

Vars.xml maps to the 'Vars' tab in the GUI. It sets up variables that can be used anywhere in the config. There may be some cases where the vairables are not being used anywhere in the config. FreeSWITCH is a young open source project and the configs are evolving and being improved over time.

cybrsrfr

Added an option to the package to disable gateways without deleting them.

cybrsrfr

A tip for those using this package. You can pickup your voicemail by dialing the extension number you are calling from. So for example if your extension is 1001 and you want to get your voicemail then from extension 1001 call 1001 and it will send you to your voicemail.

Alternative method to access voicemail is to dial extension 4000 followed by your extension number and the password.

GUI enhancements that are coming soon …

1. Management tool for adding and removing recordings
2. Management tool to setup and configure an IVR (Auto Attendant)

cybrsrfr

Minor version update. Added commands to the 'Status' tab to start, stop, restart, rescan, reloaxml and flush inbound registrations. Also found and corrected 2 minor issues with syntax.

thekod

Thanks mcrane…this package rocks.

cybrsrfr

Thanks for the encouragement.

New version is almost ready. I will try my best to get it out by sometime tomorrow (Saturday 29 Nov 2008).

scsikid

was this updated over the weekend? it looks like it wasnt based on my packages list.

is there another way to check other than the packages list page within my pfsense?

cybrsrfr

Sorry I missed my goal for weekend release still working on it. Should be soon.

scsikid

Any idea when this will be coming along Mcrane? My Linksy 941 clips the packet size of RTP and apparently there is a fix for this in the latest version of freeswitch. My phone in usable until this gets updated.

Thanks, and sorry if im sounding pushy!

cybrsrfr

I will be releasing this today Friday 5th December 2008. When its ready I will announce it here.

cybrsrfr

Announcing the release of the new pfSense FreeSWITCH package version 0.3.2.

The new package includes
1. Recordings
2. IVR (Auto Attendant)
3. 'Public' Tab
4. 'Dialplan' Tab
5. 'Gateways' Tab now has an advanced tab that hides options that are not used often simplifying the interface. Also has a tool that will make it easy to quickly add outbound dialplan rules for 7,10, and 11 digit dialing.
6. Improved 'status' Tab
Added a download button for 'call detail records' in csv format and for logs.
Added better formatting for active calls and channels.
7. New build of FreeSWITCH revision 10638 from 5 December 2008.

In the previous versions after adding a new gateway, extension or some other change the configuration was automatically reloaded. However this took a little more extra time after each change. 0.3.2 no longer reloads the configuration automatically. To reload the configuration go to the 'Status' tab and click on 'reloadxml' if you adding a gateway then press rescan. Update a gateway then press 'restart' on the external profile.

Added some documentation and links to the interface hopefully making things a bit easier.

The 'Public' tab in the previous version was a text area that allowed you to modify the XML manually. This is still available under 'deprecated xml'. Please move your config to the new public GUI interface as the 'deprecated xml' will be available for a short period of time allowing anyone who has an older install a chance to move the config.

Public tab routes inbound calls to the desired location.

Example 'Public' config in the new interface:
Extension Name: inbound_did
Enabled: true
Order: 001
Description: Inbound DID

Directs specific calls that are calling into the following DID.
Tag: Condition
Type: destination_number
Data: ^(12081231234)$

Call is transferred to extension 1001. Can also use this to direct calls to the IVR.
Tag: Action
Type: transfer
Data: 1001 XML default

–-- Upgrade Warning ------------
If you are upgrading then backup your recordings and voicemail before starting. To be absolutely you have everything backup the /usr/local/freeswitch directory. You can do this by using the 'Command' tool under 'Diagnostics' to tar gzip the directory and then use copy the file to the /tmp directory or use the 'Download' to save a copy of the file.
---- Upgrade Warning ------------

If you have any questions or problems please post a message here.

Best Regards,
Mark J Crane

cybrsrfr

Last night (8 December 2008) fixed a bug or two that prevented hearing the prompts for recording on the 'Rec' tab.
Today (9 December 2008) fixed an issue with the auto attendant (IVR) saving correctly and the audio from playing.

If you have installed the package make sure to back up /usr/local/freeswitch and then reinstall to the latest version.

thekod

I reinstalled the package to upgrade to the latest version and ended up with some missing files…the IVR tab and Gateway tab both gave 404s. Uninstalled and reinstalled the package, and it works now. So, if you run into problems, just try uninstalling and reinstalling.