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

tester_02

I'm having issues with inbound calling. Anyone have suggestions?

I've got 2 providers (2 DID), and unknowingly, I have both with the same username.
The problem is that during inbound calls, the destination_number is coming in as my username in freeswitch (both providers). So for my call routing in the public tab, I cannot use destination_number to tell which call is coming in (number is not being recognised). I thought I would use the Network_addr as a condition, but if I put the ip address, it does not find a match, even though it looks like it's right
Regex: [1001] network_addr(66.51.127.173) =~ /(66.51.127.173)/

The numbers are identical, so it has to be a formatting thing.

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.

Love the software, lots of fun things to play with still….

cybrsrfr

chpalmer: The snapshots I tested where saved on a laptop that has a power issue and will no longer power up.

tester_02: on the public tab use an anti-action as a catch all then for the action set it to info and leave the data field empty. The go to the console and watch for info a call coming in from each gateway. 'info' will send information to the console which will show in green. You can then use any of that information with the condition and direct your calls accordingly. So all you will need to do is look for something unique among the info to determine how to route the call.

On your voicemail issue it sounds like you are having a one way sound audio problem. You should test by calling another extension inside the same network. It could be simply settings needing to be adjusted on the phone.

tester_02

@mcrane:

tester_02: on the public tab use an anti-action as a catch all then for the action set it to info and leave the data field empty. The go to the console and watch for info a call coming in from each gateway. 'info' will send information to the console which will show in green. You can then use any of that information with the condition and direct your calls accordingly. So all you will need to do is look for something unique among the info to determine how to route the call.

The only unique thing that I could get was the ip.
so I set the network_addr to be equal to the ip. That did not work, so I pit the ip to be ^(ip)$ but it does not work. I've looked at some freeswitch forums but it's not leading me to the right direction, I keep getting..
Regex: [1001] network_addr(66.51.127.173) =~ /(66.51.127.173)/

It has to just be a syntax thing. Any idea what the proper sytax could be?

As for the voicemail issue…'ll play with the phone and see what's going on. Strange thing is I can have a conversation on the phone (so I know audio works), it's just the voicemail recorder that does not hear it.

One last question, what do you have for dialplan settings on your phone? Do you blank them out and let freeswitch do the plan?

Thanks again!

chpalmer

Thanks mcrane… I dropped back to 1.2.3 for now. Its working for me.

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

cybrsrfr

pfSense FreeSWITCH package 0.8.3.3

Trimmed whitespaces from freeswith.xml file and other xml files so that FreeSWITCH will now install on pfSense 2.0.

Note:
The latest pfSense 2.0 snapshot now use PHP 5.2.9. This new version of PHP uses an updated libxml which should fix the XML problems that have been experienced in pfSense 2.0.

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?

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...