1. This site uses cookies. By continuing to use this site, you are agreeing to our use of cookies. Learn More.
  2. Check out the 6 new Certified Incredible PBX Builds for Asterisk 11 and 13 featuring CentOS 6, Ubuntu 14, Raspberry Pi 2, and Asterisk-NOW.
    Dismiss Notice

TRY THIS Anyone have experience with siproxd?

Discussion in 'Help' started by bobkoure, Apr 4, 2014.

  1. bobkoure

    bobkoure Member

    When I send an inbound call to a PSTN extension (my mobile phone, part of a ring group) and pick up the PSTN, the call connects but I get no audio.
    I'm about 98% sure this is a double NAT issue, as the PBX is behind a firewall.
    The firewall does have siproxd, which works as a sip proxy at the firewall - so no NAT at all.

    I'm wondering what I have to change in the PBX to make this work.
    As best as I can figure
    asterisk SIP settings: NAT set to "never"

    In my vitelity trunks (vitelity has you set up separate outbound and inbound trunks)
    outboundproxy=[firewall LAN address]

    What am I missing?
  2. lgaetz

    lgaetz Pundit

  3. bobkoure

    bobkoure Member

    Unfortunately progressinband=yes didn't change anything.

    I made the above changes to use siproxd. Both vitelity trunks show as online. Inbound calls work fine. Outbound calls don't connect. Siproxd log isn't much help. Sigh
  4. Hyksos

    Hyksos Guru

    You should undo all that and first try to route those calls to anything that will send audio toward the inbound call, before the call is forwarded.
    IVR, annoucement, anything that will open audio toward the inbound call before the forward.
    That was mentionned also in lgaetz link also.

    As far as double NAT, nothing you mentionned points to that. You said your PBX is behind NAT, ok but why jump to saying it's double NATed? If your PBX is behind 2 NAT, why did you do that? and what is the second NAT device?
  5. bobkoure

    bobkoure Member

    Setup an IAX DID with voip.ms. Had the same problem, but progressinband=yes made it work.
    I'm sure you're right re audio; for the SIP Vitelity DID progressinband=yes didn't fix things. I'll test playing a sound via IVR when I get home (parent in hospital).

Share This Page