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    Asterisk ON pfSense2.0.1

    Scheduled Pinned Locked Moved NAT
    34 Posts 3 Posters 23.7k Views
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    • marcellocM
      marcelloc
      last edited by

      So, It's working as expected(sip gateway/proxy/server) with no audio issues?

      Treinamentos de Elite: http://sys-squad.com

      Help a community developer! ;D

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      • R
        robi
        last edited by

        @marcelloc:

        So, It's working as expected(sip gateway/proxy/server) with no audio issues?

        Running with no issues so far for 4 days:

        • gateway/router/nat
        • OpenVPN server and client simultaneously
        • asterisk

        I also plan running Snort, but I need to upgrade RAM on the box first…

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        • R
          robi
          last edited by

          I wrote a small php status page integrating in pfSense's webGUI, for anyone interested.

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          • R
            robi
            last edited by

            Some tweaks, to make Asterisk 1.8.8.1 run smoothly on pfSense box. In my case, it's a nanobsd-system, but they should work just as well on normal systems.

            /usr/local/etc/asterisk/asterisk.conf:

            [directories]
            astetcdir => /usr/local/etc/asterisk
            astmoddir => /usr/local/lib/asterisk/modules
            astvarlibdir => /usr/local/share/asterisk
            astdbdir => /tmp
            astkeydir => /usr/local/share/asterisk
            astdatadir => /usr/local/share/asterisk
            astagidir => /usr/local/share/asterisk/agi-bin
            astspooldir => /tmp
            astrundir => /var/run/asterisk
            astlogdir => /var/log/asterisk
            

            Make sure to create all the directories above if they don't exist, they are needed in order to run properly. astlogdir contains the base path where the call logs go, if that's a read-only location, simply create a symlink cdr-csv which points to a writable directory, in my case /tmp is just fine (I don't mind if I loose the file while rebooting). So it means that asterisk will look at the factory path /var/log/asterisk/cdr-csv/Master.csv actually in /tmp/Master.csv. Main asterisk log file usually goes also in astlogdir but that can be overridden below:

            /usr/local/etc/asterisk/logger.conf by the end of the file:

            console => notice,warning,error
            ;console => notice,warning,error,debug
            ;messages => notice,warning,error
            /tmp/log_asterisk => notice,warning,error
            

            Where /tmp/log_asterisk is the logfile. Note that in my status_asterisk_log.php page it looks for this file so if you change location, don't forget to update the php script too.

            /usr/local/etc/asterisk/modules.conf:

            noload => res_ael_share.so
            noload => res_adsi.so
            noload => res_agi.so
            noload => res_calendar.so
            noload => res_crypto.so
            ;noload => res_fax.so
            noload => res_jabber.so
            noload => res_monitor.so
            ;noload => res_stun_monitor.so
            noload => res_smdi.so
            noload => res_speech.so
            noload => res_odbc.so
            noload => res_musiconhold.so
            noload => app_celgenuserevent.so
            ;noload => app_confbridge.so
            ;noload => app_minivm.so
            ;noload => app_originate.so
            ;noload => app_playtones.so
            ;noload => app_readexten.so
            ;noload => app_waituntil.so
            ;noload => bridge_builtin_features.so
            ;noload => bridge_multiplexed.so
            ;noload => bridge_simple.so
            ;noload => bridge_softmix.so
            noload => cdr_adaptive_odbc.so
            noload => chan_jingle.so
            ;noload => chan_bridge.so
            noload => chan_unistim.so
            ;noload => codec_g722.so
            ;noload => format_g719.so
            noload => format_sln16.so
            noload => format_siren14.so
            noload => format_siren7.so
            ;noload => func_aes.so
            ;noload => func_audiohookinherit.so
            ;noload => func_blacklist.so
            ;noload => func_config.so
            ;noload => func_devstate.so
            ;noload => func_dialgroup.so
            ;noload => func_dialplan.so
            ;noload => func_extstate.so
            ;noload => func_iconv.so
            ;noload => func_lock.so
            ;noload => func_module.so
            ;noload => func_shell.so
            ;noload => func_speex.so
            ;noload => func_sprintf.so
            ;noload => func_sysinfo.so
            ;noload => func_version.so
            ;noload => res_curl.so
            noload => func_vmcount.so
            noload => func_volume.so
            noload => res_clialiases.so
            noload => res_config_curl.so
            noload => res_config_ldap.so
            noload => res_config_sqlite.so
            ;noload => res_limit.so
            ;noload => res_phoneprov.so
            noload => res_realtime.so
            noload => res_timing_pthread.so
            ;noload => app_adsiprog.so
            ;noload => app_alarmreceiver.so
            ;noload => app_amd.so
            ;noload => app_authenticate.so
            ;noload => app_cdr.so
            ;noload => app_chanisavail.so
            ;noload => app_channelredirect.so
            ;noload => app_chanspy.so
            ;noload => app_controlplayback.so
            noload => app_db.so
            ;noload => app_dial.so
            ;noload => app_dictate.so
            ;noload => app_directed_pickup.so
            ;noload => app_directory.so
            ;noload => app_disa.so
            ;noload => app_dumpchan.so
            ;noload => app_echo.so
            ;noload => app_exec.so
            ;noload => app_externalivr.so
            ;noload => app_festival.so
            ;noload => app_followme.so
            ;noload => app_forkcdr.so
            ;noload => app_getcpeid.so
            ;noload => app_ices.so
            ;noload => app_image.so
            ;noload => app_macro.so
            ;noload => app_milliwatt.so
            ;noload => app_mixmonitor.so
            ;noload => app_mp3.so
            ;noload => app_morsecode.so
            ;noload => app_nbscat.so
            ;noload => app_parkandannounce.so
            ;noload => app_playback.so
            ;noload => app_privacy.so
            ;noload => app_queue.so
            ;noload => app_read.so
            ;noload => app_readfile.so
            ;noload => app_record.so
            ;noload => app_sayunixtime.so
            ;noload => app_senddtmf.so
            ;noload => app_sendtext.so
            ;noload => app_setcallerid.so
            ;noload => app_sms.so
            ;noload => app_softhangup.so
            noload => app_speech_utils.so
            ;noload => app_stack.so
            ;noload => app_system.so
            ;noload => app_talkdetect.so
            ;noload => app_test.so
            ;noload => app_transfer.so
            ;noload => app_url.so
            ;noload => app_userevent.so
            ;noload => app_verbose.so
            ;noload => app_voicemail.so
            ;noload => app_waitforring.so
            ;noload => app_waitforsilence.so
            ;noload => app_while.so
            ;noload => app_zapateller.so
            ;noload => cdr_csv.so
            noload => cdr_custom.so
            ;noload => cdr_manager.so
            noload => cdr_pgsql.so
            noload => cdr_radius.so
            noload => cdr_sqlite.so
            noload => cdr_sqlite3_custom.so
            noload => cdr_syslog.so
            ;noload => cel_custom.so
            ;noload => cel_manager.so
            noload => cel_odbc.so
            noload => cel_pgsql.so
            noload => cel_radius.so
            noload => cel_sqlite3_custom.so
            noload => cel_tds.so
            ;noload => chan_agent.so
            noload => chan_gtalk.so
            noload => chan_iax2.so
            ;noload => chan_local.so
            ;noload => chan_mgcp.so
            ;noload => chan_multicast_rtp.so
            noload => chan_oss.so
            ;noload => chan_sip.so
            noload => chan_skinny.so
            ;noload => codec_a_mu.so
            ;noload => codec_adpcm.so
            ;noload => codec_alaw.so
            ;noload => codec_g726.so
            ;noload => codec_gsm.so
            ;noload => codec_lpc10.so
            ;noload => codec_speex.so
            ;noload => codec_ulaw.so
            ;noload => format_g723.so
            ;noload => format_g726.so
            ;noload => format_g729.so
            ;noload => format_gsm.so
            ;noload => format_h263.so
            ;noload => format_h264.so
            ;noload => format_ilbc.so
            noload => format_jpeg.so
            ;noload => format_ogg_vorbis.so
            ;noload => format_pcm.so
            ;noload => format_sln.so
            ;noload => format_vox.so
            ;noload => format_wav.so
            ;noload => format_wav_gsm.so
            ;noload => func_base64.so
            ;noload => func_callcompletion.so
            ;noload => func_callerid.so
            ;noload => func_cdr.so
            ;noload => func_channel.so
            ;noload => func_curl.so
            ;noload => func_cut.so
            noload => func_db.so
            ;noload => func_enum.so
            ;noload => func_env.so
            ;noload => func_frame_trace.so
            ;noload => func_global.so
            ;noload => func_groupcount.so
            ;noload => func_logic.so
            ;noload => func_math.so
            ;noload => func_md5.so
            noload => func_odbc.so
            ;noload => func_pitchshift.so
            ;noload => func_rand.so
            ;noload => func_realtime.so
            ;noload => func_sha1.so
            ;noload => func_srv.so
            ;noload => func_strings.so
            ;noload => func_timeout.so
            ;noload => func_uri.so
            noload => pbx_ael.so
            ;noload => pbx_config.so
            ;noload => pbx_dundi.so
            ;noload => pbx_loopback.so
            ;noload => pbx_realtime.so
            ;noload => pbx_spool.so
            ;noload => res_clioriginate.so
            noload => res_config_pgsql.so
            ;noload => res_convert.so
            ;noload => res_mutestream.so
            ;noload => res_rtp_asterisk.so
            ;noload => res_rtp_multicast.so
            ;noload => res_security_log.so
            ;noload => res_snmp.so
            noload => cdr_odbc.so
            noload => cdr_tds.so
            noload => chan_h323.so
            noload => res_config_odbc.so
            

            Tons of modules there, the ones uncommented in the config file will not be loaded by Asterisk. I don't use these anyways, so less memory usage, less conflicts and no errors at all with this setup.

            /usr/local/etc/asterisk/sip.conf the most important one:

            [general]
            alwaysauthreject=yes
            language=hu
            maxexpiry=600
            defaultexpiry=100
            registerattempts=250
            registertimeout=15
            allowguest = no
            bindport=5060
            bindaddr=0.0.0.0
            localnet=192.168.1.0/255.255.255.0
            localnet=192.168.2.0/255.255.255.0
            externhost=your.domain.name.can.be.dyndns.too
            externrefresh=600
            jbenable=yes
            disallow=all
            allow=g729
            allow=ulaw
            allow=alaw
            
            

            If you have multiple LAN subnets, add them all here as localnet, if SIP phones are on them. It's important for security reasons.
            Note bindaddr=0.0.0.0, that means Asterisk will listen directly on all interfaces of pfSense box. That means no firewall configuration is needed at all (no rules, no nat etc): clients inside LAN and servers out there will be able to talk to each other through Asterisk, nothing needs to be routed anywhere.

            /usr/local/etc/asterisk/extensions.conf requires nothing special, just configure it as you wish.

            Look in your logfile periodically. If Asterisk complains about something, a Google search on the error message usually throws back usable suggestions. Any security issues may show up in the logs too if it's the case.

            Good luck!

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            • marcellocM
              marcelloc
              last edited by

              I can help you on first package relase.

              It's basically a XML file That install php files, create menus and install the asterisk files.

              Isn't better start a topic on packages section instead of nat?

              Treinamentos de Elite: http://sys-squad.com

              Help a community developer! ;D

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              • R
                robi
                last edited by

                A moderator please move the topic in the right place.

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                • R
                  robi
                  last edited by

                  @marcelloc:

                  I can help you on first package relase.

                  It's basically a XML file That install php files, create menus and install the asterisk files.

                  How to do that?

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                  • marcellocM
                    marcelloc
                    last edited by

                    take a look on pfsense-packages at github.com

                    https://github.com/bsdperimeter/pfsense-packages

                    I'll create the xml file to install your php files.

                    Treinamentos de Elite: http://sys-squad.com

                    Help a community developer! ;D

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                    • B
                      blietaer
                      last edited by

                      Sweeeet!
                      I think you have achieved more stability that I coul din my original hack/post, cheers!
                      Maybe 2.0.1 helps, or you VoIP provider is better/closer and/or you directory redirections help….
                      I'll give a shot to these improvements.

                      But most important for me, using a very poor country side 3M/128k DSL line, I really have to enable QoS in order to shut down/lower any non-VoIP traffic while a call is taking place....any input? Wizards seemed of no use for me...

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                      • R
                        robi
                        last edited by

                        @blietaer:

                        But most important for me, using a very poor country side 3M/128k DSL line, I really have to enable QoS in order to shut down/lower any non-VoIP traffic while a call is taking place….any input? Wizards seemed of no use for me...

                        In sip.conf, general section add this:

                        [general]
                        disallow=all
                        allow=g729
                        

                        Make sure to keep the correct order, disallow first, allow the second.

                        And of course check your SIP phones and your VoIP provider, do they support g729 codec (Linksys/Cisco and most SIP phones support it).
                        g729 is actually mpeg-compressed audio in 8kpbs, while alaw and ulaw are uncompressed pcm at 64kbps - sound quality is the same. Don't use alaw and ulaw unless you have plenty of bandwidth.

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                        • R
                          robi
                          last edited by

                          @marcelloc:

                          I'll create the xml file to install your php files.

                          Thanks! I'll post a small update.

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                          • B
                            blietaer
                            last edited by

                            Concerning the /usr/local/etc/asterisk/modules.conf, you really mean this is the whole content, or do you suggest to add your snipset at the end of any existing modules.conf file ?
                            same for the others ? (thouI guess you Sip.conf _is_longer than that…)

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                            • R
                              robi
                              last edited by

                              I posted only parts of the config files, which are affected by my modifications.

                              In modules.conf, yes: just add what I posted at the end of the file.
                              In sip.conf I posted my general section, which should be at the beginning of each file. The other sections are each dependent of the particular setup, because every section defines a separate SIP client account (for a SIP phone device, for example).

                              There's no point thus, to post my entire config files, they are mostly irrelevant for others. What I suggested, are changes which actually make the thing work at all on pfSense nanobsd.

                              But if one looks into asterisk's documentation, everything will become straight clear.

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                              • marcellocM
                                marcelloc
                                last edited by

                                First package release is out.  :)

                                http://forum.pfsense.org/index.php/topic,47210.msg248054.html#msg248054

                                I did some changes to improve stability and checks for nanobsd or normal install.

                                Treinamentos de Elite: http://sys-squad.com

                                Help a community developer! ;D

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