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

chpalmer

Just a report:

My provider is getting this line from me now (instead of the standard sip:1234567890@myipaddy:5080;transport=udp)

sip:gw+Voip Line 1@myipaddy:5080;transport=udp

pfsense 1.2.3 package 0.8.3.1

Was playing some more last night. "The Voip Line 1" comes from my gateway name. Dont know where the gw+ comes from. Is there a way to fix this?

cybrsrfr

In order for me or anyone else to help you please provide more information on your configuration.

Or try IRC freenode at: #pfsense-freeswitch

chpalmer

Sorry…

1.2.3-PRERELEASE-TESTING-VERSION
built on Tue Mar 3 22:30:29 EST 2009

package now... 0.8.3.3

This is a fresh install of both the firewall and Freeswitch just to make sure...

The config I was using worked up till the last two packages. Ptretty much default otherwise.

Reinstall- Ive tried starting over without any backups from configs.

Provider is pointed back to my gateway name "Voip Line 1" and not my user id. "1231231234"

Anything else I can tell you?

Im lost...

jflaugh

It looks like this topic is the place to ask questions so here it goes.
My issue is I installed a Vertical Xcelerator IP and I want to use the IP phones outside of my network I thought this would be a simple task but it is not with SIP.
I was told by Vertical that UDP/TCP port 5060 and UDP port range 35000 to 65000 needs forwarded and 1:1 NAT needs setup. I looked into 1:1 NAT but the from what I understand I need a second static ip and that is not possible. So in my quest to pull this off I found FreeSWITCH with the proxy option.

What I want to accomplish… Put my SIP/IP/VOIP PBX behind my PFsence and have external hard phone connection + have the PBX connect to a SIP provider. Can FreeSWITCH help me.

Here is my network setup
PFsence 1.2.2
external 207.255.XXX.110/24
internal 192.168.78.1/24
Ip PBX 192.168.78.30

If there is anyone that could advise me on the best solution to get my SIP out on the inernet I would greatly appreciate the help.
Thanks
Jeremy

thekod

If you've got the server internal, and the phones external, just forward the ports and you should be fine…you shouldn't need 1:1 or FS proxy, unless I'm missing something.

jflaugh

I think the issue I am having is RTP and NAT no playing nicely I need someting to accept and forward the RTP packets that NAT is not expecting.

thekod

Forwarding rules may do this. I had success doing this another way, see http://forum.pfsense.org/index.php/topic,12830.0.html, but I don't know if pieces of that may work for you…

Taras_

I have fresh install of pfSense 1.2.2 & fresh install of FreeSWITCH 0.8.3.4.
FreeSWITCH don't start.

Here details:


# cd /usr/local/freeswitch/bin/
# ./freeswitch
Cannot lock pid file /usr/local/freeswitch/log/freeswitch.pid.
# rm /usr/local/freeswitch/log/freeswitch.pid
# ./freeswitch
Cannot lock pid file /usr/local/freeswitch/log/freeswitch.pid.
# ps aux | grep freeswitch
#

cybrsrfr

Sorry uploaded the new build but forgot to rename and replace the old one. Remove the package and reload it and you should get the new version that will work.

Mark

0tt0

Hi,

Just for the record: FreeSWITCH is now running here (hasn't been under the latest 2-3 versions).
Am using 0.8.3.4 on pfsense 1.2.2

Cheers,

chpalmer

@chpalmer:

@chpalmer:

Just a report:

My provider is getting this line from me now (instead of the standard sip:1234567890@myipaddy:5080;transport=udp)

sip:gw+Voip Line 1@myipaddy:5080;transport=udp

pfsense 1.2.3 package 0.8.3.1

Was playing some more last night. "The Voip Line 1" comes from my gateway name. Dont know where the gw+ comes from. Is there a way to fix this?

Found this regarding my issue. http://www.nabble.com/Contact-header-on-INVITE-td22376608.html

In case anyone else runs into it… :)

iced98lx

The 0.8.3.4 package is 1.2.2 safe with no need for PHP work around. It also fixes the recording glitch where 2 seconds of recording got tagged on the end. Easy upgrade no issues, great work Mcrane!

Chris

phdonnelly

@tester_02:

I also have an issue where if I call in and the calls goes to voicemail, I cannot leave a message. As I talk into the phone to leave the message, it stops and says it is below minimum record length, even thought I am still talking. This is not an issue if I am calling from extension to extension.

Hey, I had the same exact issue with voicemail, just though I'd let you know, placing an "Answer" action immediately before the voicemail action fixed things right up for me. It's odd because it was working before without it, but it's an easy enough fix.

tekoholic

Although I am not sure, it doesn't seem as if anyone's aware of the issues in the recent several 2.0AA snapshots, with installing FreeSwitch. Each time I've tried to install it, it hangs at:

Configuring package components…
Additional files... libcurl.so.5

At this point, the whole web-gui hangs, reboot results in errors regarding curl (specifically: PHP Warning: PHP Startup: Unable to load dynamic library '/usr/local/lib/php/20060613/curl.so' - /usr/local/lib/libcurl.so.5: invalid file format in Unknown on line 0), nothing else works correctly until removal of both of these files and a reload of the 2.0AA full update...

Not sure if this is best posted here, or on the 2.0AA Problems support section, but it seems to be the only package having this sort of trouble...

edit: for the addition of error details...

cybrsrfr

pfsense 2.0 package installer seems to have a problem with requesting remote files using libcurl. So for those that really want it I have created a manual install. That is php code that can be run from Diagnostics -> Command -> PHP Execute.

The manual install php code is attached to this forum post but you have to be logged into the forum to see it.

pfsense_2.0_manual_install.txt

tekoholic

@mcrane:

pfsense 2.0 package installer seems to have a problem with requesting remote files using libcurl. So for those that really want it I have created a manual install. That is php code that can be run from Diagnostics -> Command -> PHP Execute.

The manual install php code is attached to this forum post but you have to be logged into the forum to see it.

Mcrane is SO far beyond a Hero Member!! The code, attached to his previous post, worked wonders getting FS running on the newest snap of 2.0AA.

What kinda' steaks do ya' like with yer' brew, mcrane? We'll make a BBQ of it!!

adrianhensler

Some more questions.

Is anyone using CallForwarding? I'm not sure it's working for me. That is to say, it is certainly not working but I'm not sure if it is the phone (Grandstream 2020 and Linksys 962) or if I am doing something wrong. Is it possible to enable with a code like "*72"

Also - DISA - is there something precanned to get this to work or should I just be looking at the freeswitch.org examples?

This example on freeswitch wiki looks good to me but I haven't tried it yet:

http://wiki.freeswitch.org/wiki/Examples_disa.js

Thanks for any opinions.

cybrsrfr

@adrianhensler:

Is anyone using CallForwarding? I'm not sure it's working for me. That is to say, it is certainly not working but I'm not sure if it is the phone (Grandstream 2020 and Linksys 962) or if I am doing something wrong. Is it possible to enable with a code like "*72"

When I've done it found that on the Internal profile if aggressive nat detection is enabled it messes up call forwarding to other extensions in some situations. Even with it disabled certain situations call forwarding to an extension has directed the call directly to voice mail. You could troubleshoot the problem using /usr/local/freeswitch/bin/./fs_cli and try to get additional help from #freeswitch IRC channel. Or I have also created a followme.js (simultaneous or in succession) file that could be used to direct calls to more than one number. I'm planning on extending it soon so that you can dial from an extension and get numbers that are chosen through the phone's dial pad. At this moment the file would have to manually be changed. http://wiki.freeswitch.org/wiki/Examples_followme.js

@adrianhensler:

Some more questions.
Also - DISA - is there something precanned to get this to work or should I just be looking at the freeswitch.org examples?

This example on freeswitch wiki looks good to me but I haven't tried it yet:

http://wiki.freeswitch.org/wiki/Examples_disa.js

Thanks for any opinions.

I wrote that example DISA you linked to so my opinion about it is likely biased. Tonight (technically morning now) I updated the DISA example with a few improvements, improved the instructions on the FreeSWITCH wiki, and then added a link to the recordings that are needed for the DISA script.

Hope that helps.

adrianhensler

"Hope that helps" he says.

You did pretty much everything other than sign in and configure it for me!

I'll let you know how I make out.

Thanks!

michaelahess

Real n00b questions here but I can't get x-lite to register, all I get is:

Registration error: 408 - Request Timeout

I've got this in X-Lite:

Display name: FreeSwitch
User name: 2001
Password: 2001
Authorization user name: 2001
Domain: 10.x.x.1 (pfSense box)

Checked Register with domain

What I've done on the pfSense box is:

Set domain var to 10.x.x.1
Added extension 2001 with above info
Changed Default Area Code (not trying to go to an external service, just internal between clients right now)
The only dialplan is Recordings

Everything under status tab says Disconnected, goodbye….

Service says it's running.

I've no idea what to do at this point.