Speech Server audio problems with Asterisk RRS feed

  • Question

  • I've managed to have the following setup:
    VoIP provider <=> Asterisk 1.6.1 Beta 1 <=> Speech Server 2007

    Asterisk is currently hosted in a VM (PBX-in-a-Flash 1.3) on the same box as Speech Server as we are evaluating the use of Speech Server to develop voice response applications. I can successfully place a call from outside (cell phone) and hear the voice response application but sending DTMF tones and audio doesn't work at all. Within our LAN, softphones can connect and interact properly with the voice response application.

    The VoIP provider mentioned that RTP packets are not reaching our system. After a bit of investigation it seems that Asterisk behind a NAT is plagued with all sorts of these problems: one-way audio, no audio, etc. I'm wondering if there is a better solution than using Asterisk as the middle man; it would definitely be better if the solution was Windows-based as well and that it would handle the TCP<=>UDP issues.

    Are there any options besides Asterisk that might be easier to setup?
    Friday, October 10, 2008 4:50 PM

All replies

  • Try freeSwitch.  Some people are reporting success with it.  But it won't solve the NAT problem.  The server needs to be on the internet with a public IP or I don't think it's possible, unless you have a SIP-aware router/firewall (expensive!).


    Friday, October 10, 2008 5:57 PM
  • I did download and compile freeswitch. I'll be looking into it again next week.

    However it seems like a lot to ask to setup a machine on the public IP because then you'd still need a software-based firewall for security.  The Asterisk VM was using iptables internally anyway.

    What kind of SIP-aware router/firewalls are available on the market? Isn't SIP and RTP just another protocol on top of UDP and in that case any router that can forward traffic from an IP should suffice (although that is one of the problems I'm seeing right now... the router seems to be dropping RTP packets).
    Friday, October 10, 2008 7:19 PM
  • I couldn't get Freeswitch configured. It seems like I would have to spend more time as the documentation on it is not as extensive.

    However, I got lucky doing a search on Google. I found this article that explained some configuration settings:
    I basically had some settings in sip_custom.conf that needed to go in sip_general_custom.conf and some settings were missing. The only thing different was that I didn't use externhost setting.

    Now 2-way audio works perfectly fine.
    Tuesday, October 14, 2008 8:04 PM
  • Hi, could you please give us a little detail on how configured the Asterisk to route calls to the Speech Server? I'm trying to do this exactly same thing (Provider -> Asterisk -> Speech Server) but documentation is hard to find.
    I'm using Asterisk and FreePBX 2.6, so it would be very nice to be able con configure everything through the UI (freePBX).

    Thanks a lot,
    Thursday, February 4, 2010 3:20 PM
  • I very interesting in route call from Asterisk to the Speech Server too.
    I find this site http://gotspeech.net/blogs/michaeldunn/archive/2007/08/28/asterisk-speech-server-2007.aspx but not work for me.
    any help will be appreciated
    Tuesday, February 9, 2010 4:52 PM