[FreeSWITCH] All audio not reaching server

Guiyon

Trying to setup my first FreeSWITCH install and I'm running into a bit of problem. I am using PFSense 1.2.3-RC3 and the regular FreeSWITCH package with X-Lite 4.0 Beta (Mac) as the client. The problem I am running into is that I can hear audio from the server fine (ie: the *9998 hold music) but no audio (including DTMF tones) are detected server-side. Both the echo test (*9996) and voicemail recording fails. At the moment FreeSWITCH is configured using the default settings and just a single extension has been added and it has been setup to bind to the LAN address (instead of the WAN). If you need any other information let me know and I'll post it ASAP.

Ritmo2k

I just encountered this yesterday as well…
I don't yet know enough about FS to make an educated comment on the complete validity of the article at http://wiki.freeswitch.org/wiki/Multi_home_tutorial but it states when using a multihomed setup you must make some changes to the binding ip's for the sip profiles and the directory/default.xml file in the http://wiki.freeswitch.org/wiki/Multi_home_tutorial#GETTING_YOUR_LAN_PHONES_REGISTERED section. Every time I made that change it broke my system completely. I ended up leaving the value as "$${domain}" and changing vars.xml to:


  <x-pre-process cmd="set" data="domain=10.0.0.1"><- Your LAN int ip</x-pre-process>

hth

cybrsrfr

It would help to know a little more information on the your network layout.
For example is FreeSWITCH running on your pfSense that is on the perimeter firewall or is it running on a seperate box on the inside of your network.

Another question what phone are you using. xlite at least running on windows handles NAT very well so it is a good one to test with. Linksys phones handle NAT okay depends on how they are setup.

Also I suggest going to pfSense packages and removing siproxd from the machine that is running FreeSWITCH.

Also go to NAT setings and set advanced outbound NAT settings. It will create automatically generated rules edit them so they show static to yes. By default pfSense does this for port 5060 for SIP but FreeSWITCH uses 5060 through 5090 for SIP.

Additional information can be found here:
http://doc.pfsense.org/index.php/FreeSWITCH

inksite

Hi,

I'm also having this problem. I just recently deployed pfsense on a dedicated PC in order to support a dual-wan (single-lan) environment. Freeswitch will use a Gateway on only the primary WAN connection; in fact, that seems to be working well on a test account I've set up with Flowroute. I'm testing with X-Lite. It seems no outbound audio (microphone or DTMF) is reaching the server. I cannot get past the pin prompt to record voicemail messages. I can successfully call a POTS phone on the PSTN through Flowroute and hear inbound audio, but the recipient cannot hear any audio including DTMF from the X-Lite client. Per this thread, I made sure that siproxd package was not installed and I set the outbound NAT as

Any additional suggestions would be a great help.

Thanks.

Michael Keen
http://www.inksite.com

Follow up: I was successful in setting up an X-Lite client at a remote WAN location with full audio by setting up port forwarding to the LAN address of the pfsense/freeswitch computer. But I cannot manage to resolve the one-way audio issue on the LAN side PC's. Still hoping for a reply here. Thanks.

cybrsrfr

I helped inksite over Skype. These are the changes we did to get this to work:

1. Remove NAT entries that pushed VOIP somewhere else on the network.
2. Register the phones to the WAN ip or domain name pointed to the WAN IP
3. Delete the lan profile. LAN profile can be used but it makes its harder to get everything working. Its far easier to force the internel profile to use the LAN IP then it is to use a seperate profile.