Odd Issue CentOS 7 Vs Ubuntu 20.04

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So I have CentOS 7 server running asterisk 18.8.0 – all is good.

I unplug that server – plug in a ubuntu 20.04 server at the same IP address. let my 3 devices reconnect to the ubuntu server….

When I pick up the polycom phone and dial it connects. I hear the other ends ‘tone” – but when I press digits – nothing happens
(to select a port)
Seems everything is set for rfc2833.

The devices are a TOA SIP Gateway, and a TOA N-8000 device connected to the GW.

I have compared the settings of the polycom extension on both boxes – they match and also the SIP gateway.

I tried to compare the sip debug from the Ubuntu to the CentOS and “looked”
the same to me.

Where might I look next or what might I look at ?

Thanks,

Jerry

5 thoughts on - Odd Issue CentOS 7 Vs Ubuntu 20.04

  • ok – if I “rtp set debug on ” on the CentOS 7 server I get a tone of logging.

    if I do the same on the ubuntu 20.04 all i get is like 2 lines. I have done “systemctl stop firewalld” on the ubuntu box – same result.

    Where do I look next ?

    Jerry

  • I dont get it – I certainly getting RTP traffic because I defined an extension to playback the demo-congrats messages. I call that extension – and ALL kinds of RTP traffic prints on teh console.

    But when I call the one extension – 103 – all it prints is 2 lines.

    I also removed the source tree – un tarred – ran the contrib/scripts/install_prereq install script, it did install a couple packages – I dont think they mattered. do the ./configure, make, make install and started up again – same issue though.

    Jerry

  • So – still on this…

    I was just dialing the SIP Gateway with Dial(SIP/103)

    if I change my Dial command to this:

    Dial(SIP/103,20,D(15))
    So I send out the DTMF in the dial command – this works and connects me and the DTMF is delivered and I get the right port.

    The problem still remains – Dialing just Dial(SIP/103) from the polycom phone – and then doing 15 for DTMF does not work. Cant figure out why ?

    Any thoughts ?

    Jerry

  • The usage of D(15) causes Asterisk to produce RTP on its own. Without it, it merely forwards RTP. If a NAT/firewall requires media to be sent before allowing media in, then you’ll have no media flow. You can use the
    “rtpkeepalive” option to have the RTP stack produce keepalive packets, which will then open the NAT/firewall.

  • This ended up being a simple canreinvite situation… I had yes – and needed to be set to NO. Jerry