Netgate Discussion Forum
    • Categories
    • Recent
    • Tags
    • Popular
    • Users
    • Search
    • Register
    • Login

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

    Scheduled Pinned Locked Moved pfSense Packages
    314 Posts 39 Posters 286.5k Views
    Loading More Posts
    • Oldest to Newest
    • Newest to Oldest
    • Most Votes
    Reply
    • Reply as topic
    Log in to reply
    This topic has been deleted. Only users with topic management privileges can see it.
    • chpalmerC
      chpalmer
      last edited by

      Thanks:  I think thats my problem. It appears the adapter may not be specifying the ports to the pbx correctly.

      http://zoom.com/  These guys have been around for years. I used to have a 33k modem made by them. (Now Im not sure I actually have a modem in my house anywhere, but thats for another post…)

      Im setting up another ata right now as I post this to seperate them out for more controlled conditions...

      :) ;D

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

      1 Reply Last reply Reply Quote 0
      • C
        cybrsrfr
        last edited by

        For those with a linksys pap2 each phone port has a different port by default example line 1 5060 and line 2 5061. I don't have a zoom yet but I believe it would be the same.

        What model of zoom are you using?  I've been looking at one zoom that has both 1 fxs and 1 fxo. The feature that really caught my eye was ilbc support.  Ilbc codec is comparable to G729 if not better with its audio compression.

        1 Reply Last reply Reply Quote 0
        • chpalmerC
          chpalmer
          last edited by

          I have a model 5822 that was never released fully by them. I was a beta tester for a voip company and they sent it to me. I can find only one setting for local sip port on it. I believe it will only be good for one port.

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

          1 Reply Last reply Reply Quote 0
          • 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.

            1 Reply Last reply Reply Quote 0
            • 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

              1 Reply Last reply Reply Quote 0
              • 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.

                1 Reply Last reply Reply Quote 0
                • C
                  cybrsrfr
                  last edited by

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

                  1 Reply Last reply Reply Quote 0
                  • 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."

                    1 Reply Last reply Reply Quote 0
                    • 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>

                      1 Reply Last reply Reply Quote 0
                      • 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

                        1 Reply Last reply Reply Quote 0
                        • 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.

                          1 Reply Last reply Reply Quote 0
                          • 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");

                            1 Reply Last reply Reply Quote 0
                            • 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 
                              
                              1 Reply Last reply Reply Quote 0
                              • 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.

                                1 Reply Last reply Reply Quote 0
                                • 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.

                                  1 Reply Last reply Reply Quote 0
                                  • 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.

                                    1 Reply Last reply Reply Quote 0
                                    • 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.

                                      1 Reply Last reply Reply Quote 0
                                      • T
                                        Taras_
                                        last edited by

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

                                        1 Reply Last reply Reply Quote 0
                                        • 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]

                                          1 Reply Last reply Reply Quote 0
                                          • 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.

                                            1 Reply Last reply Reply Quote 0
                                            • First post
                                              Last post
                                            Copyright 2025 Rubicon Communications LLC (Netgate). All rights reserved.