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

    Scheduled Pinned Locked Moved pfSense Packages
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    • C
      cybrsrfr
      last edited by

      FreeSWITCH package minor update.

      Add International option to the Dialplan Expression tool,
      Minor wording changes
      Hide Outbound Caller ID on the Extensions tab. Effective Caller ID is used by FreeSWITCH.

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      • C
        cybrsrfr
        last edited by

        New FreeSWITCH build 1.0.2 revision 11245. More modules have been compiled including mod_fax, mod_openzap, and mod_spidermonkey_odbc.

        More info can be found on mod_fax at:
        http://wiki.freeswitch.org/wiki/Mod_fax

        In order to use fs_cli you would connect with ssh, then run the following commands. Replace the 192.168.1.1 with your own pfSense LAN IP address. The port and the password are settings that can be configured from the 'Settings' tab.

        cd /usr/local/freeswitch/bin
        ./fs_cli -H 192.168.1.1 -P 8021 -p ClueCon

        The following is a list of modules that are available. # is a comment any module with this in front of it was not compiled.

        loggers/mod_console
        loggers/mod_logfile
        loggers/mod_syslog
        applications/mod_commands
        applications/mod_conference
        applications/mod_dptools
        applications/mod_enum
        applications/mod_fifo
        #applications/mod_fax (trying to build this time: failed)
        applications/mod_voicemail
        #applications/mod_lcr
        applications/mod_limit
        applications/mod_expr
        applications/mod_esf
        #applications/mod_easyroute
        applications/mod_fsv
        applications/mod_soundtouch
        applications/mod_rss
        applications/mod_snom
        applications/mod_vmd
        asr_tts/mod_flite
        asr_tts/mod_pocketsphinx
        #asr_tts/mod_cepstral
        codecs/mod_g723_1
        codecs/mod_amr
        codecs/mod_g729
        codecs/mod_h26x
        codecs/mod_voipcodecs
        codecs/mod_ilbc
        codecs/mod_speex
        codecs/mod_siren
        codecs/mod_celt
        dialplans/mod_dialplan_directory
        dialplans/mod_dialplan_xml
        dialplans/mod_dialplan_asterisk
        directories/mod_ldap
        endpoints/mod_dingaling
        endpoints/mod_iax
        #endpoints/mod_portaudio
        endpoints/mod_sofia
        endpoints/mod_loopback
        #endpoints/mod_alsa
        #endpoints/mod_opal
        ../../libs/openzap/mod_openzap
        event_handlers/mod_event_multicast
        event_handlers/mod_event_socket
        event_handlers/mod_cdr_csv
        #event_handlers/mod_radius_cdr
        formats/mod_native_file
        formats/mod_sndfile
        formats/mod_shout
        formats/mod_local_stream
        formats/mod_tone_stream
        #languages/mod_python
        languages/mod_spidermonkey
        languages/mod_spidermonkey_teletone
        languages/mod_spidermonkey_core_db
        languages/mod_spidermonkey_socket
        languages/mod_spidermonkey_odbc
        languages/mod_lua
        #languages/mod_perl
        #languages/mod_yaml
        xml_int/mod_xml_rpc
        xml_int/mod_xml_curl
        xml_int/mod_xml_cdr
        #xml_int/mod_xml_ldap
        say/mod_say_en
        say/mod_say_de
        say/mod_say_es
        say/mod_say_fr
        say/mod_say_it
        say/mod_say_nl
        say/mod_say_zh

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        • A
          adrianhensler
          last edited by

          mcrane: You are putting features in faster than I can play with them.

          What's the status on the mod_openzap - if I put a digium card or clone in my machine, will my chances of making it work be reasonable?  I have an ebay X100 type clone.

          I took a look at the freeswitch wiki and it looks promising, is everything in place in the pfsense package to make this work? Or do I have to manually get the zaptel OS level modules in pfsense?  Sorry, seems like a newbie question - but to get a zaptel card in my pfsense box I have to remove the wireless or move pfsense to a different box due to having only the one PCI slot.

          I'd rather not go through those steps if it's not something that will work anyways - and I'm tempted to just use all voip anyways and drop the analog line.  Still up for debate.  Can't wait to look at the fax module.

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          • C
            cybrsrfr
            last edited by

            I'm testing mod_openzap right now. I will report the results here.

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            • M
              mrguitar
              last edited by

              ….I can't wait to play w/ the mod_fax.

              I vote that we change mcrane's status from Hero Member to "Super Hero Member."

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              • C
                cybrsrfr
                last edited by

                mrguitar: I'm glad the work is appreciated! :)

                Here is a little more…

                SIP (TLS/SSL) and RTP (SRTP) Encryption Notes

                FreeSWITCH

                Run the following from SSH or the Console
                    ./gentls_cert setup
                    //replace freeswitch.org with your domain name or IP address
                    ./gentls_cert create -cn freeswitch.org -alt DNS:freeswitch.org

                Vars tab

                <x-pre-process cmd="set" data="sip_tls_version=sslv23">Enable SSL for the Internal profile

                <x-pre-process cmd="set" data="internal_ssl_enable=true"><x-pre-process cmd="set" data="external_ssl_enable=false">Dialplans tab (default.xml)
                    In order to encrypt both sides of the call
                    Uncomment the following line:
                      <action application="export" data="sip_secure_media=true"><condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$" break="never"><action application="set" data="sip_secure_media=true"></action></condition>

                Status tab
                    Should show (TLS) if it is enabled.
                    internal profile   sip:mod_sofia@67.60.128.195:5061 RUNNING (0) (TLS)
                    internal-ipv6 profile           sip:mod_sofia@[::1]:5061 RUNNING (0) (TLS)

                Linksys SPA942
                  SIP tab
                    SRTP Method: s-descriptor

                Ext 1
                    SIP Transport: TLS
                    SIP Port: 5061
                    SRTP Private Key: leave blank

                Can make the calls secure using activation codes or set from the 'Users' tab.

                Users Tab
                      Secure Call Setting: Yes

                Regional (Activation Codes ) 
                      Secure All Call Act Code: *16
                      Secure No Call Act Code: *17
                      Secure One Call Act Code: *18
                      Secure One Call Deact Code: *19

                For Additional information see:

                FreeSWITCH Wiki
                  SIP (TLS / SSL)
                  http://wiki.freeswitch.org/wiki/Tls#Linksys_TLS_Setup

                SRTP
                  http://wiki.freeswitch.org/wiki/SRTP</action></x-pre-process></x-pre-process></x-pre-process>

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                • M
                  mrguitar
                  last edited by

                  mcrane,

                  I might have found a small bug. (sorry if this has already been discussed, I didn't want to re-read this whole thread again)

                  I uploaded a lot of recordings and setup several IVRs. I noticed that none of the sounds would "stick" in the IVR config screen unless I removed the "Auto" name and renamed them. Once the files were named, the IVRs worked great.
                  Cheers,
                  mrguitar

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                  • S
                    scsikid
                    last edited by

                    Anytime I try to uninstall the freeswitch package, it seems to hang and never uninstalls. I installed the latest version 0.6 and I think I interrupted the install. Now I cant uninstall it. I simply get the following and it just hangs there.

                    
                    Removing package... 
                    Loading package configuration freeswitch.xml... 
                    Loading package instructions...
                    

                    Any help would be appreciated.

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                    • C
                      cybrsrfr
                      last edited by

                      @scsikid:

                      Anytime I try to uninstall the freeswitch package, it seems to hang and never uninstalls. I installed the latest version 0.6 and I think I interrupted the install. Now I cant uninstall it. I simply get the following and it just hangs there.

                      
                      Removing package... 
                      Loading package configuration freeswitch.xml... 
                      Loading package instructions...
                      

                      Any help would be appreciated.

                      Scroll down to the bottom of the page and see if there are any errors. If there are then please report the error.

                      Manual steps to remove the FreeSWITCH package run the following.

                      pfSense GUI -> Diagnostics -> Command -> PHP Execute

                      exec("killall -9 freeswitch");
                      unlink_if_exists("/usr/local/pkg/freeswitch.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch.inc");
                      unlink_if_exists("/usr/local/pkg/freeswitch_dialplan.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch_extensions.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch_external.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch_internal.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch_modules.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch_public.xml");
                      unlink_if_exists("/usr/local/pkg/freeswitch_vars.xml");
                      exec("rm -R /usr/local/freeswitch/");
                      exec("rm -R /usr/local/www/freeswitch/");
                      unlink_if_exists("/usr/local/etc/rc.d/freeswitch.sh");
                      unlink_if_exists("/tmp/freeswitch.tar.gz");
                      unlink_if_exists("/tmp/pkg_mgr_FreeSWITCH.log");

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                      • S
                        scsikid
                        last edited by

                        Didnt even see that error down there.

                        Warning: delete_package(/usr/local/pkg/freeswitch.inc): failed to open stream: No such file or directory in /etc/inc/pkg-utils.inc on line 740 Fatal error: delete_package(): Failed opening required '/usr/local/pkg/freeswitch.inc' (include_path='.:/etc/inc:/usr/local/www:/usr/local/captiveportal:/usr/local/pkg') in /etc/inc/pkg-utils.inc on line 740 
                        
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                        • C
                          cybrsrfr
                          last edited by

                          The error says that it can't find the file /usr/local/pkg/freeswitch.inc which is the include that defines the steps to uninstall the package. The FreeSWITCH package will not work if the freeswitch.inc file is missing. Without that file you will definitely need to run manually remove the package as I described in a previously.

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                          • C
                            cybrsrfr
                            last edited by

                            @mrguitar:

                            I might have found a small bug. (sorry if this has already been discussed, I didn't want to re-read this whole thread again)

                            The bug had not been reported.

                            @mrguitar:

                            I uploaded a lot of recordings and setup several IVRs. I noticed that none of the sounds would "stick" in the IVR config screen unless I removed the "Auto" name and renamed them. Once the files were named, the IVRs worked great.
                            Cheers,
                            mrguitar

                            Actually the act of editing the recording no changes and then saving it would fix the problem. In looking into this I found another bug where the guid would change each time the edit page was saved. I have made some change that are available in 0.6.3 that should fix these problems. Please confirm that it is working now.

                            Anyone using IVR/Recordings is strongly encouraged to upgrade to 0.6.3 or higher.

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                            • chpalmerC
                              chpalmer
                              last edited by

                              Actually the act of editing the recording no changes and then saving it would fix the problem. I looking into this I found another bug where the guid would change each time the edit page was saved. I have made some change that are available in 0.6.3 that should fix these problems. Please confirm that it is working now.

                              Could this be a reason my recordings are unusable after upgrading packages?

                              Thanks for all the hard work Mcrane! ;D :)

                              Triggering snowflakes one by one..
                              Intel(R) Core(TM) i5-4590T CPU @ 2.00GHz on an M400 WG box.

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                              • C
                                cybrsrfr
                                last edited by

                                @chpalmer:

                                Could this be a reason my recordings are unusable after upgrading packages?

                                Yes, after upgrading the package it would have also re-generated the guid which would have broken the IVR link to the recordings. To fix it you would have needed to select the recordings again under the IVR tab. This bug should be fixed. If you get a chance please confirm that this bug is fixed.

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                                • T
                                  Taras_
                                  last edited by

                                  Thanks for excellent work!
                                  Now I'm testing FreeSWITCH with two VoIP providers: Sipnet (out) and Telphin (in & out).

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                                  • T
                                    Taras_
                                    last edited by

                                    Have one problem: can't send voicemail to email.

                                    My settings:
                                    SMTP host: my.smtp.server:465
                                    SMTP auth: false
                                    SMTP secure: no
                                    SMTP from: pbx@pfsense.my.domain

                                    On my.smtp.server in maillog found this (related to my pfsense's IP address):
                                    Jan 24 18:25:03 www postfix/smtpd[10206]: connect from unknown[xx.xx.252.153]
                                    Jan 24 18:25:03 www postfix/smtpd[10206]: lost connection after RSET from unknown[xx.xx.252.153]
                                    Jan 24 18:25:03 www postfix/smtpd[10206]: disconnect from unknown[xx.xx.252.153]

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                                    • C
                                      cybrsrfr
                                      last edited by

                                      If you have your own email server to use to send the mail.

                                      Port :465 is for TLS which I'm guess you are not using so your settings would be something like the following. Second thing to keep in mind is that to send email to the mail server without smtp authentication you will have to allow mail relay on your mail server for your pfsense ip address. Mail relay is usually disabled to prevent spammers from using your mail server to send spam.

                                      SMTP host: my.smtp.server
                                      SMTP auth: false
                                      SMTP secure: no
                                      SMTP from: pbx@pfsense.my.domain

                                      The Gmail is a simple example for those that have a Gmail account but don't necessarily have their own mail server.
                                      http://doc.pfsense.org/index.php/FreeSWITCH#Voicemail_to_Email

                                      SMTP Host: smtp.gmail.com:465
                                      SMTP Secure: tls
                                      SMTP Auth: true
                                      SMTP Username: Use your gmail email address here.
                                      SMTP Password: Use your gmail email password here.
                                      SMTP From: Use your gmail email address here. It may support any valid email address but this has not been tested.
                                      SMTP From Name: Can be anything you choose. For my example I used: voicemail.

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                                      • T
                                        Taras_
                                        last edited by

                                        All of requirements are ok. I'm able to send mail via telnet my.smtp.server 465 without any problem (while connecting to SMTP server from my pfsense's ip):

                                        telnet my.smtp.server 465
                                        ehlo myhostname
                                        mail from: pbx@pfsense.my.domain
                                        rcpt to: <my extension's="" email="">data
                                        Subject: test
                                        test

                                        I'll try different SMTP server soon.</my>

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                                        • T
                                          Taras_
                                          last edited by

                                          I have same problem with another mail server.

                                          Also FreeSWITCH profile external don't starts after pfsense reboot (stop/start of profile helps), may be because of my WAN link is PPTP?

                                          Here my /tmp/voicemailtoemail.txt after trying to send mail:

                                          
                                          # cat /tmp/voicemailtoemail.txt
                                          To: taras@1adm.ru
                                          From: "Lucheev Dima" <102@10.10.15.1>
                                          Subject: Voicemail from "Lucheev Dima" <102> 00:00:11
                                          
                                          Mailer Error: Language string failed to load: connect_host
                                          
                                          
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                                          • C
                                            cybrsrfr
                                            last edited by

                                            Taras Savchuk: contact me on irc #pfsense-freeswitch my username is mcrane and we can look further into the 'Language string failed to load' error.

                                            @Taras:

                                            Also FreeSWITCH profile external don't starts after pfsense reboot (stop/start of profile helps), may be because of my WAN link is PPTP?

                                            Yes most likely if FreeSWITCH external loads before the PPTP interface connects then the external profile fails to start. Would need to look into a way to detect that and start it if it is not running. This could be done with some custom PHP code and the new pfSense package I wrote called 'PHP Service.'

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