Problem with Freeswitch package
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I'm not a freeswitch user yet, still in asterisk with chan_datacard.
There isn't a log file just like asterisk /var/log/asterisk/full?
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Scratch that, jinxed by kartweel.
Sorry! :)
If you disable packet filtering on the pfSense box does it then work?
I found the easiest way to see what was happening was doing packet captures on the various interfaces to see where the data was going.
It is positive to know you had it working for a little while :)
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OK, did a bit of testing with Freeswitch and FusionPBX with FusionPBX at version 1877.
With packet filtering set to 'on' aka normal operation of PFSense 2.01 x386:
Dialing internal extension from another internal extension results in the destination phone not ringing and the call timing out to voice-mail. From the dialing extension, there is no ring back tone nor does the dialing extension hear the voice mail prompt when the dialed extension times out to voice mail.With packet filtering set to 'off' in PFSense 2.01 x386:
Dialing an internal extension from another internal extension: Destination phone rings however when answered, audio is only one-way where audio is only being sent from the handset that placed the call. You can hear audio from the originating handset coming through the destination handset but there's no audio returning from the destination handset to the originating handset. Echo test *9196 works although the music on hold *9664 does not result in any audio but appears to be spooling the audio file based on the freeswitch log contents.Conclusion:
FreeSwitch w/FusionPBX works fine on pfsense 2.0 x86 however something must have changed with how SIP traffic is routed between PFSense versions 2 and 2.01 as this seems to be a plausible explanation for the one way audio. I cannot see any errors in either pfsense logging or Freeswitch logging that would indicate a possible avenue to pursue although someone with more experience in freeswitch behavior might be able to spot something amiss.I think I might revert back to 2.0 x86 and rebuild the freeswitch and fusionPBX along with svn to reaffirm the full operation under 2.0 but though I would at least post findings with 2.01 as a cautionary tale for those entertaining moving a production critical pfsense 2.0 with Freeswitch to pfsense 2.01.
Shaun
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Sip again, :-\
Take a look on this topic to see if you have same issue
http://forum.pfsense.org/index.php/topic,45255.msg236346.html#msg236346
There maybe some missing ldd to freeswitch libs, freeradius had this bootup issue with mysql libs fixed in package reload script
http://forum.pfsense.org/index.php/topic,43675.msg233938.html#msg233938
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It could very well be the same issue that N8LBV is encountering with his Asterisk setup. I have a different configuration with a different SIP PBX system but the symptoms appear to be the same. I wonder if he's having any issues with internal to internal calling?
Here's my setup:
1. PFSense 2.01 x86 on a Nano-ITX board with two built in ethernet interfaces.
2. WAN interface is set to DHCP from the ISP.
3. LAN interface is static and set to 192.168.1.1 address.
4. DHCP is enabled on LAN interface and DNS forwarding is 'checked' to allow internal DHCP clients to use the ISP's DNS.
SIP phones on the 192.168.1.0/24 have static addresses. //Nothing was changed on the phones when moving from PFSense 1.2.3 to 2.01. The PFsense 2.01 was a fresh install and reconfig from scratch.
5. External SIP provider configured with a single 'channel' or phone number.
Note: registration with SIP provider works fine and outbound calls originating from any of the internal SIP hard-phones always works and has two way audio as it should.DynDNS service is enabled and the DynDNS public DNS name record is used on the phones to force them to attach to the WAN interface and not the LAN interface. //This logic is leftover from Freeswitch on PFSense 1.2.3 as I think the Freeswitch 'package' for PFSense 1.2.3 was configured to bind on the WAN interface and not the LAN…not sure why this was the case other than it would make things a bit easier if using an external SIP gateway or allowing external SIP phones to connect to the internal Freeswitch system.
Let me know if you would like me to try anything. I have the weekend to experiment and test before the pfsense box takes on it's weekday production role.
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The installation before reboot was working?
if so, can you re-install freeswitch package and its dependencies.
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It appears that 'working' prior to the reboot would have been a relative term. The functions that I primarily use are outbound calling via the SIP gateway provider which does work and continues to work from any handset on the LAN side. I didn't start trying handset to handset calling until after rebooting from our svn installation effort and likely falsely assumed it to be working given the outbound calling success…it's what I get for being so optimistic. All of the internal handsets are able to register via SIP with Freeswitch and the displays are all accurate showing the correct number of missed calls and all dial directly but there's no audio from the destination handset for any internal to internal handset call.
I'm going to perform a clean install of PFSense 2.0 from an ISO I still have and then perform the Freeswitch and FusionPBX install per Mark's instructions again, confirm operation and then perform a subsequent clean install of PFSense 2.01. I don't think I ever fully tested Freeswitch and FusionPBX on 2.0, certainly not to the depth I have with PFSsense 2.01.
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Here's the latest.
Fresh install of PFsense 2.0 x86 and following the Freeswitch and FusionPBX install process outlined in the wiki at http://wiki.fusionpbx.com/index.php/PfSense_Install
1. Internal handset to handset no audio and no ringing.
1a. Turn packet filtering off and ringing works however there's only one way audio. Same symptoms as the original configuration on PFsense 2.01.
2. SIP registration works.
3. Outbound SIP calls work, inbound SIP calls are not passed through.Fresh install of PFSense 2.01 x86 and FusionPBX:
Same exact symptoms as with PFSense 2.0.
Same problem as Kartwheel with no audio.Shaun
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Are you listening freeswitch on all interfaces?
Is there on freeswitch config some kind of nat settings you could disable?
One last thing, can you try this setup with asterisk?
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By default Freeswitch is binding to the WAN interface which is how it was setup on PFSense 1.2.3 with the Freeswitch 'package' installed…and worked. I see the point of trying to have it bind to the LAN interface address but I'm not sure how to do that but will look into it.
You mentioned Asterisk, is it possible to slip Asterisk onto a PFSense 2.0.1 system?
Getting a bit tired of banging my head on the Freeswitch/FusionPBX on PFSense 2.01 method.
I added our little one way SIP audio issue to the FusionPBX forum as a final gasping effort. http://www.fusionpbx.com/messageboard/rsssub2list.php?rssid=6&rsssubid=206 there doesn't seem to be much activity on the site meaning expectations of insight or response are fairly low.I did see this as well: http://www.fusionpbx.com/messageboard/rsssub2list.php?rssid=6&rsssubid=193 but it doesn't address LAN to LAN SIP one way audio issue.
I can't shake the idea that the root of the issue seems to be related to the difference in how PFSense 2.x deals with SIP via NAT (LAN-WAN-LAN) compared to how it was handled in PFSense 1.2.3.
Thanks,
Shaun -
test asterisk
amd64
pkg_add -r http://e-sac.siteseguro.ws/packages/amd64/8/All/asterisk18-1.8.8.1.tbzi386
pkg_add -r http://e-sac.siteseguro.ws/packages/8/All/asterisk18-1.8.8.1.tbzThis doc may be usefull too:
http://www.freepbx.org/forum/freepbx/tips-and-tricks/asterisknow-freepbx-and-pfsense
