SIPROXD and Asterisk / Trixbox

  • Has anyone been successful getting SIPROXD to work with Asterisk or Trixbox? If so how did you them?



  • I would love some info. as i use Elastix ( same thing as Trixbox ) & would love to know the benefit of using SIPROXD.

  • I ahve been having problems with one-way audio. I have tried everything I can think of with pfSense including static ports but nothing seems to help.

    The issue with one-way audio is that the PBX is behind the firewall in a NAT'd environment so when the packets come back if the inbound port on the firewall changes audio becomes one-way. This can be fixed by specifying a STUN server but Asterisk doesn't support it.

    SIPROXD fixes the issue by relaying the packets from and to the PBX from the firewall. Since the firewall is on the public network port changes aren't an issue. If the SIPROXD implementation is transparent (meaning the PBX doesn't know anything about it and the firewall redirects the packets automatically to the SIPROXD app) things just work.

    I have been unable to get SIPROXD working on pfSense with the Trixbox (Asterisk) PBX. I believe the issue is that the packets for port 5060 are not being redirected by pfSense to the SIPROXD app. This is failrly straightforward in Linux with iptables but I'm afraid my BSD knowledge is too limited to see if this is the issue… let alone fix it.


  • SIProxyD would be nice but I believe it has been broke for quite some time. However I have gotten SIP to work behind pfSense using Asterisk with several different providers so it definitely can be done with the right settings on Asterisk and pfSense. In fact I have one of my phones behind a second pfSense firewall inside my internal network and it also works with sound both directions.

    Did you make the changes to the sip.conf file I suggested in the following post?,8682.msg50287.html#msg50287

  • Actually I have had the changes you suggested loaded for quite awhile but I have still been getting one way audio.

  • I'm a little confused. I am running pbxinaflash with two sip trunks, one with Telasip and the other with stanaphone, both work flawlessly with no port forwarding or siproxd?

  • I wish I knew what the difference is… I am planning to convert to "PBX in a Flash" soon to see if it will fix issues I am having with poor sound quality on IAX calls but I get occasional one-way audio with SIP and pfSense. the interesting thing is that the one-way audio goes away after about 10 seconds and it works for a little bit and then I get it again. The 10 seconds appears to be related to the externrefresh=10 setting.

    BTW... I am using voipstreet for SIP and IAX but I had the same one-way audio problems with Broadvoice a year or so ago.


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