FreeSWITCH package for pfSense 1.2.1 and 2.0 released. PBX or Proxy
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I tested pfSense on the pfSense 1.2.3 snapshot it is working fine so far. But I should note that I had to disable SIP TLS support to get the 'Internal profile' to start. pfSense 1.2.3 is based on FreeBSD 7.1. I believe to get SIP TLS (used to encrypt SIP) to work with FreeBSD 7.1 will require a specific TLS build for FreeBSD 7.1.
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Thanks mcrane! :)
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4 things - 1. I AM IMPRESSED!!! Awesome work MCRANE
2. One little thing I noticed that didnt seem to be the appropriate behavior:
I made a gateway and entered a dialplan expression for that gateway and saved it. When I was looking at the settings in that gateway again the dialplan expression was blank, so I thought, hmm must have forgot it, and added it again and saved.
When I go to the Dialplan tab it now has a duplicate of the dialplan I just built for that gateway, but looking at the gateway again it is blank again - is this how it should work or a bug, or am I just expecting too much ;?
3. I am running this on the new pf vmware image, and I attempted to configure a gateway to my asterisk box on the same LAN, but when I do this and watch the asterisk console, nothing ever happens, and freeswitch just says FAIL_WAIT. I have played with the settings a bunch, to no avail, I am using a freepbx extension, but at any rate, I should see some failure on my asterisk cli at 50 verbosity. Is there an example of this that I can use? I saw this http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk but was hoping to use a freepbx extension.
4. I didnt notice a way to specify the port for a particular gateway - it would be nice if that were available on the "Gateways" page, I am assuming its something you set on the "External" page.
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4 things - 1. I AM IMPRESSED!!! Awesome work MCRANE
I'm glad you appreciate it.
2. One little thing I noticed that didnt seem to be the appropriate behavior:
I made a gateway and entered a dialplan expression for that gateway and saved it. When I was looking at the settings in that gateway again the dialplan expression was blank, so I thought, hmm must have forgot it, and added it again and saved.
When I go to the Dialplan tab it now has a duplicate of the dialplan I just built for that gateway, but looking at the gateway again it is blank again - is this how it should work or a bug, or am I just expecting too much ;?
It is working how it was designed. The 'Dialplan Expression' tool under the Gateway tab is simply a shortcut for adding an outbound route to the 'Dialplan.' As stated on that page: "Shortcut to create the outbound dialplan entries for this Gateway. The entries are saved to and edited from the 'Dialplan' tab. "
3. I am running this on the new pf vmware image, and I attempted to configure a gateway to my asterisk box on the same LAN, but when I do this and watch the asterisk console, nothing ever happens, and freeswitch just says FAIL_WAIT. I have played with the settings a bunch, to no avail, I am using a freepbx extension, but at any rate, I should see some failure on my asterisk cli at 50 verbosity. Is there an example of this that I can use? I saw this http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk but was hoping to use a freepbx extension.
Currently I don't have an Asterisk box to experiment with what you are trying to do. I migrated all my installations of Asterisk over to FreeSWITCH a couple of months ago.
The link you are referencing is the link I would have suggested to you. However it may help to know that by default FreeSWITCH binds to your WAN IP. You might need some rules or NATing to get it directed to your asterisk box that is on the LAN. It might be easier to have FreeSWITCH bound to the LAN and then it should have no problem talking to asterisk that is located on the LAN. To do that look at the instructions here: http://doc.pfsense.org/index.php/FreeSWITCH#IP_address_or_Domain
4. I didnt notice a way to specify the port for a particular gateway - it would be nice if that were available on the "Gateways" page, I am assuming its something you set on the "External" page.
If the my domain for my gateway was proxy.mygateway.com you should be able to adjust the port with proxy.mygateway.com:5062.
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A new FreeSWITCH update is available that adds ability for an extension to have more than one voicemail email address assigned to it. All you need to do is seperate the email addresses by a comma or a semi-colon. This would allow you to send your voicemail to as many email addresses as desired. You can even send it to a special email address for your cell phone provider so you will get a text message notification of your voicemail.
Reminder: voicemail can also be accessed by calling your extension number or by calling extension 4000 then providing your extension number and voicemail password.
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Works perfectly! Thanks mcrane.
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ok, thanks for the advice. keep in mind i am using the new vmware image, which by defualt has allow any/any to/from wan - wan is bridged to lan, unless i am misunderstanding what was stated on the blog, and the firewall rules shown on my box, but I am guessing you are right in that i need to mess with the ports or bind fs to the lan.
on a sidethought - i noticed there is a Snom module, which I like very much, but my utmost preferred phone is aastra for their huge xml apps, is there any thought to include them? I would think the native xml of FS would only compliment aastra phones (better than asterisk;) with their xml and php addons. I can help in porting the aastra package except for the dialplan components, which I will most likely not be fully comfortable with for a while, so - ya I am willing to do what ever small part in this area, and as much testing as anyone needs with aastra phones. something helpful- http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-D521E521/03/hs.xsl/23492.htm
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on a sidethought - i noticed there is a Snom module, which I like very much, but my utmost preferred phone is aastra for their huge xml apps, is there any thought to include them? I would think the native xml of FS would only compliment aastra phones (better than asterisk;) with their xml and php addons. I can help in porting the aastra package except for the dialplan components, which I will most likely not be fully comfortable with for a while, so - ya I am willing to do what ever small part in this area, and as much testing as anyone needs with aastra phones. something helpful- http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-D521E521/03/hs.xsl/23492.htm
I haven't played much with asstra and don't currently have any for development. You are welcome to help out in any way you desire.
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New version available for the TFTP package.
1. The new version automatically adds the TFTP option to the DHCP Server.
2. Configuration files in the /tftpboot directory can now be modified by clicking on the edit button. Make your changes and then click on 'Save.'== Note ==
If you have any problems with the TFTP double check that the tftp lines in the /etc/inetd.conf are not commented out.
The beginning of the line looks like
TFTP is commented out and will not run.
#tftp dgram udpTFTP is ready to go.
tftp dgram udp -
I got a weird issue with any recordings be it voice mail or IVR… as the recording ends the last several syllables repeat. Its happened for a while and Im wondering if something on my box is corupt or maybe a setting I can change.
I am running on pfSense 2.0 at this point. pfSense-Full-Update-2.0-ALPHA-ALPHA-20090115-2300.tgz Newest version as of yesterday seemed to break the freeswitch package.
Latest freeswitch package also but this has been happening for a couple of weeks.
Anyway just curious.. :)
Im using a Grandstream 502 here without any issues. Dialing with RFC28*** doesn't work. Using sip info. (normal?)
Dlink DVG 1402s wont stop ringing after it gets a call... (latest firmware)
Zoom 5822 works but seems to loose registration intermittently. (only good for one port)
:) ;D
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but my utmost preferred phone is aastra for their huge xml apps, is there any thought to include them? I would think the native xml of FS would only compliment aastra phones (better than asterisk;) with their xml and php addons.
Totalimpact: I spent an awfully long time - hours - trying to get an aastra to register against my pf[reeSwitch] box. Have you been successful? I had this same aastra registered against an asterisk server; but could not get it to register at all to the freeswitch. My linksys 962 took all of two minutes to get working; and that includes getting the Slashdot rss feeds on the phone.
I might have been doing something wrong, so if you get an aastra working I'd be interested in seeing screenshots of your config screens.
I was using a aastra 9133i with the latest firmware.
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no Adrian - i havent actually tried to get one going on it, I will see if I can check one out tomorrow. Aastras are pretty finicky with network and nat settings - you are connecting everything on the same network - right? I still have not had a chance to troubleshoot why FS wont register to an asterisk extension, but I am pretty sure its something in the network config.
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Ok, so I got a 57i plugged in, it registers, but it fails to make calls. everything is on the local lan, and this phone worked fine yesterday on asterisk. So i did a packet capture, and the packets looks like this:
1. SIP/SDP Request: INVITE
2. SIP Status: 100 Trying
3. SIP Status: 407 Proxy Authentication Required
4. SIP Request: ACK sip:9000@192.168.2.123:5060
5. IP Fragmented IP protocol (proto=UDP 0x11, off=0) [Reassembled in #6]
6. SIP/SDP Request: INVITE
7. same as packet 5I dont know if this Aastra specific, or freeswitch, but something keeps breaking the invite packets, they get reassembled in the next packet, but keep breaking. Softphones work fine, I even tried a different switch. At any rate I opened a ticket with aastra support, i will see if they can make anything of it.
@Adrian - How far did you get with your 9133?
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It may help to monitor freeswitch you can use fs_cli (command line interface)
SSH to pfsense FreeSWITCH machine then:
cd /usr/local/freeswitch/bin
./fs_cli -H 192.168.0.1Replace 192.168.0.1 with pfsense LAN ip address.
Then you can watch what is going on in FreeSWITCH.
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ya - thanks, I was just messing around at the cli. I am hoping these will work since they are listed on the FS interop page.
on another note - my Nokia N80 over wifi works flawless, actually better than the softphone;).
yet another: I cant figure out why this gateway that I deleted keeps popping up on the Status page under "sofia status" - is there more than one place I need to delete a gateway? I even tried rebooting.
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Hi,
short question. Where can I find the current release of your freeswitch package?
Within the packages tab I can see only release 0.7 of freeswitch…thanks in advance
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Within the packages tab I can see only release 0.7 of freeswitch…
That's the latest one as far as I know.
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Hello mcrane…. I've finally setup another pfsense box to test out freeswitch. I'm a complete newb with a pap2(currently running 1 line with v-buzzer), but want to venture and try out a small SOHO setup.
I'm reading and reading, but I'm having a hard time with the initial setup. Is there a chance someone has a walkthrough for just the initial setup. I am sure that after I can get the system to at least talk to a phone and get the basics working, the rest is easy. The only way for me to figure out pfsense was to get the basic firewall working, and then one by one I learned how to setup and use the additional features (snort/proxy).
After I install the basic freeswitch package, what are the settings that have to be modified for a simple 2 phone system? How about settings for a pap2/spa942?
Funny thing was I originally thought I could do it by just some simple reading, and I was going to post a simple walkthough, as I know someone else who is interested. Instead it's me asking how..... -
Did you see this page?:
http://doc.pfsense.org/index.php/Category:FreeSWITCH -
Also Mcrane - I was able to finally get an Aastra 57i to work. evidentally it was generating packets that were exceeding the 1500 byte packet limit, so it was fragmenting the packets and for some reason pfsense/freeswitch was not able to reassemble them.
There is a fix for this on the phone: In the phone web-UI: Global SIP> Codec Preference List> Codec 1 >
change all to "Basic Codec", save settings and restart the phone.Or in cfg files for aastra set:
sip use basic codecs: 1
Not sure if there is a space for that in the wiki.
