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

adrianhensler

Some quick notes:

I meant to mention earlier - for the benefit of anyone reading this thread that the RTP packet size tip earlier in this thread is a great improvement. I had been aware of this - but it's a big difference in quality if this change is implemented.

Re: reload xml - I was pretty careful to do that every time. I was doing other stuff at the same time so may have inadvertently missed once but the results are as above - it works no matter what I put in:

<anti-action application="export" data="domain_name=${sip_req_host}"></anti-action>

Blank works( data="domain_name=" ) or ( data="domain_name=mydomainname.com" ) also works.

Does the line a few lines under that one:

 <condition field="destination_number" expression="^(10[01][0-9])$" <="" pre="">… have to be edited as well to match my extensions?  I've changed them to be in the 20xx range and edited the "local_extensions" as documented under the extensions tab.

And finally, regarding the force-register-domain...
Didn't seem to work. I uncommented it and left it as it was - I'm assuming that was the correct thing to do.

Editing the public.xml does seem to do the trick though.

I'm using 1.2.1-rc4 - are others running the 2.0 beta?  Maybe it's not picking up my domain correctly?  Again - editing public.xml seems to work so that's fine - maybe I'll figure out why my configuration is different while I look at other parts.'

Thanks everyone the help, even though my asterisk configuration was doing everything I needed, it's nice to try something different.

Now to try to fix that no-ring only static on outbound calls.  I'll try another provider first and see what results I have with that; then I'll look at trying to fix the one I am currently using.</condition>

adrianhensler

Just tested with les.net for outgoing and the call was fine.

Didn't hear the ringing of the pstn line from the sip-freeSwitch side - haven't looked into that yet. When the call was answered everything was fine; the caller could hear me very clearly but I could not hear the caller very clearly. Haven't tried anything at all to figure out why - suffice it to say that I've used these two providers for a year or more and the quality has been quite reasonable; and my cable modem is typically 15000 / 1200 for speeds.

Great work mcrane; it's not child's play to set it up but it's pretty close.

Now I have to set up my basic IVR and get some 'fake' extensions forwarding to remote lines.

Sorry - I really hijacked this thread. Didn't mean to - seemed the right place to ask questions but maybe I should have started a new thread. Keeps this one at the top anyways!

cybrsrfr

This thread can take it. A bug was found on the IVR today by 'tusc' I will get that fixed soon.

Mark

cybrsrfr

The IVR bug has been fixed.

@adrianhensler:

Great work mcrane; it's not child's play to set it up but it's pretty close.

'sabo11' volunteered to write a tutorial so that should make it easier to get started when its finished.

adrianhensler

My no-ring and outgoing calling not working with one provider (link2voip.com) are all fixed - the issue was a checkbox on my phone. Troubleshooting with a softphone let me narrow down where the problem was in a big hurry.

Haven't messed with the emailing of voicemail yet or the IVR but the package is looking like it will do everything I need to replace my asterisk box.

Very pleased, going to try to do some customizing over the next few days.

adrianhensler

Hi,

I think I found a bug.

If a gateway has a number in it - eg sip.ca1.link2voip.com - the dialing doesn't seem to work correctly.

geek.org SET [effective_caller_id_name]=[Adrian softphone]
2008-12-19 00:26:25 [DEBUG] switch_core_state_machine.c:152 switch_core_standard_on_execute() sofia/internal/2009@adrian.is-a-geek.org Execute bridge(sofia/gateway/${default_gateway}/1${default_areacode}4881718)
2008-12-19 00:26:25 [DEBUG] switch_core_session.c:1253 switch_core_session_execute_application() sofia/internal/2009@adrian.is-a-geek.org Expanded String bridge(sofia/gateway//19024881718)
2008-12-19 00:26:25 [ERR] mod_sofia.c:2322 sofia_outgoing_channel() Invalid Gateway
2008-12-19 00:26:25 [NOTICE] mod_sofia.c:2521 sofia_outgoing_channel() Close Channel N/A [CS_NEW]
2008-12-19 00:26:25 [ERR] switch_ivr_originate.c:1071 switch_ivr_originate() Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]

I took a quick look but didn't find where this might be happening.

Note the expanded: (sofia/gateway//19024881718)

You can sort of verify this by just adding a number to an existing gateway; then restart freeswitch then try to dial - you will see the string doesn't get populated as above and you get the sofia/gateway//19025551234)

Sorry if I'm off again but this is my best guess as to what is happening here.

cybrsrfr

Are you referring to the gateway name which is the first field under editing a gateway?

If you are then I changed the name of one of my gateways for testing to the name you provided sip.ca1.link2voip.com and in my test the calls worked fine and the name was not lost.

Note: If you do add a gateway or change the name make sure to 'restart' the external profile from the status tab.

thekod

adrian, is it possible you had an invalid character in the name?

adrianhensler

Thanks for the tip on restarting the external, I had been restarting the entire freeswitch service.

To be sure that I didn't have any bad characters I deleted the gateway and added it back.

The problem appears to be only when I try to add 7 digit dialing, it works with 10 digit for example.

I still get this when I try to do 7 digit dialing:

(sofia/gateway/${default_gateway}/1${default_areacode}4881718)
2008-12-19 08:54:31 [DEBUG] switch_core_session.c:1253 switch_core_session_execute_application() sofia/internal/2009@adrian.is-a-geek.org Expanded String bridge(sofia/gateway//19024881718)

Okay - deleted all the link2voip gateways. Found some left over in config.xml? Lots from my previous testing. Deleted those manually. Added it back in with 7 digit dialing. I was pretty sure it would work at this point but it did not.

Then verified it looked good to me:

 <config>
                                <dialplanincludeid>{CAB6ADA6-413F-592C-7A86-9A52359389AD}</dialplanincludeid>
                                <tag>condition</tag>
                                <fieldtype>destination_number</fieldtype>
                                <fielddata>^(\d{7})$</fielddata>
                        </config> <config><dialplanincludeid>{CAB6ADA6-413F-592C-7A86-9A52359389AD}</dialplanincludeid>
                                <tag>action</tag>
                                <fieldtype>bridge</fieldtype>
                                <fielddata>sofia/gateway/sip.ca1.link2voip.com/1902$1</fielddata></config>

What's up; it all looks fine but the log shows it is dropping the gateway somewhere.

Ahhhhh…..
Found the fix I think.

After testing, the problem only seemed to be with 7 digit dialing. Found there was a 01_example.com.xml in /usr/local/freeswitch/conf/dialplan/default with this bit of code for an example 7 digit dialing:

 <extension name="local.example.com"><condition field="${toll_allow}" expression="local"><condition field="destination_number" expression="^(\d{7})$"><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/${default_gateway}/1${default_areacode}$1"></action></action></action></condition></condition></extension>

I think this is what was tripping up my 7 digit dialing; I deleted it and everything is gravy.

Maybe I'm not understanding how some of the configs flow through; but it seems like there were some 'old' gateways still showing in the config.xml that I couldn't see in the gui as well as this example code - should I have been able to see this in the gui under the dialplan?

Thanks again for all the help. Sorry if these are newbie type mistakes on my part.

tusc

mcrane,

I've been meaning to mention this. I believe this example file should be removed from you package as it takes precedence over a user's dialplan configuation.

@adrianhensler:

After testing, the problem only seemed to be with 7 digit dialing. Found there was a 01_example.com.xml in /usr/local/freeswitch/conf/dialplan/default with this bit of code for an example 7 digit dialing:
 <extension name="local.example.com"><condition field="${toll_allow}" expression="local"><condition field="destination_number" expression="^(\d{7})$"><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/${default_gateway}/1${default_areacode}$1"></action></action></action></condition></condition></extension> 
I think this is what was tripping up my 7 digit dialing; I deleted it and everything is gravy.

Maybe I'm not understanding how some of the configs flow through; but it seems like there were some 'old' gateways still showing in the config.xml that I couldn't see in the gui as well as this example code - should I have been able to see this in the gui under the dialplan?

Thanks again for all the help. Sorry if these are newbie type mistakes on my part.

thekod

I had to modify that file to get dialing to work for me…but only 10digit. And adrianhensler, it seems that freeswitch package upgrades sometimes leave behind those entries, in my experience.

cybrsrfr

The following are changes made on 19 December 2008 in version 0.5.3. These changes remove some of the conflicting configs that are found in the default config. Clear up the public.xml setting that has made it harder for new installs. These changes clear up the configuration issues that have been discussed in the forum up to this point.

Details:

Delete the example default config files they are not needed with the GUI
/usr/local/freeswitch/conf/dialplan/default/01_example.com.xml
/usr/local/freeswitch/conf/dialplan/public/00_inbound_did.xml

Convert extensions tab from pfsense package XML code to PHP. This cleared up a problem that was showing up on pfSense 2.0.

Public tab change 'deprecated xml' to 'public.xml'. This provides access to a file that may be useful in advanced cases.

For new installs public.xml a setting that was giving trouble inside the public.xml at /usr/local/freeswitch/conf/dialplan/public.xml
has been changed from this
<anti-action application=""export"" data=""domain_name=${sip_req_host}"/">to this
-->

The setting was overriding the correct value and causing the issue reported earlier in this thread.

For those that had an earlier version than 0.5.3 you need to go to the 'public' tab then click on the 'public.xml' button. At that link you will see a text area box which should have the content of the attached file. For new installs this will auto populate for older installs the text will need to be manually copied back into the text area.

public.xml.txt</anti-action>

adrianhensler

For some reason, I had to manually copy these files over:

from the cvstrac:

exec("cp  /tmp/freeswitch_extensions.tmp  /usr/local/www/freeswitch/freeswitch_extensions.php"); 
        unlink_if_exists("/tmp/freeswitch_extensions.tmp"); 

        exec("cp  /tmp/freeswitch_extensions_edit.tmp  /usr/local/www/freeswitch/freeswitch_extensions_edit.php"); 
        unlink_if_exists("/tmp/freeswitch_extensions_edit.tmp");

I'm almost sure it's a fluke on my system only; can't see anything that would have stopped this from copying those files over. I just mention it in case someone else runs into the 404 file not found on these files.

thekod

To avoid 404s on files for this package, I always have to uninstall the package and reinstall…dunno why

cybrsrfr

Thanks to tusc there should now be mod_shout support which provides support for mp3 audio.

Also this new build of FreeSWITCH enables mod_siren, and mod_celt both exciting new codecs.

mod_celt (58kHz audio)
http://freeswitch.org/node/155

mod_siren (32 kHz and 64 kHz)
http://freeswitch.org/node/153

adrianhensler

You guys are on fire.

cybrsrfr

FreeSWITCH package 0.5.6 Fixed an issue with the mod_shout mp3 now really does work.

Music on hold directory. Can simply add mp3 or wav files to this directory to change the music on hold.
/usr/local/freeswitch/sounds/music8000

If SSH is enabled under System -> Advanced -> Enable Secure Shell then you can use SFTP to the folder above. Firefox add on FireFTP works pretty well with SFTP.

Added package info pointing to:
http://doc.pfsense.org/index.php/FreeSWITCH

Free Christmas Music.
http://feelslikechristmas.com/contemporary-instrumental-christmas-music-mp3s.html

Instrumental.
http://expelledthemovie.com/digitaldownloads.php

Political. Ron Paul theme song
http://www.aimeeallen.com/RPsong.html

cybrsrfr

Found a bug in the IVR (Auto Attendant). Its now fixed in version 0.5.7. If you will be using the IVR then please upgrade to 0.5.7 or higher.

jalgaz

mcrane, thx for your great work!

I need a litte help.
How to connect a softphone from WAN to a fs at my LAN. (soft – NAT -- WAN -- NAT -- LAN)
A B
I think, in A use stun, in B nated ports 5060-5090 and rtp port to my pfsense
In A, sjphone in htc touch pda
Rules in Wan / Lan permit traffic SIP / RTP / STUN
No firewall in OS Ubuntu

1.- i create extension 1000, 1001 in context default
2.- i create extension 1003 in context public
3.- step 1 ok, calls ok, sofphones can register, but 1003 can't register to fs
I view freeswitch wiki, but i can't find the solution
there is not much information in the web
I modify the external profile, the parameters of sip_port,... no register
I create doublenat.xml... no register
At logs no info from attempts to connect.
Any idea?

Un saludo from Spain!

adrianhensler

Here's some thoughts.

I think they should all be context of default.

Make an extension 1003 in default and try it on the lan. When you know it works; apply the same configuration to your softphone.

Also; check the logs immediately after it registers (or fails to register) and see if there are any clues there.

I hope this helps; I am sure someone else will correct me if I am wrong.