On Fri, Feb 4, 2022 at 12:42 PM Jerry Geis <jerry.g...@gmail.com> wrote:
> > > On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis <jerry.g...@gmail.com> wrote: > >> >> >> On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis <jerry.g...@gmail.com> wrote: >> >>> >>> >>> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis <jerry.g...@gmail.com> wrote: >>> >>>> 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 >>>> >>> >>> >>> 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 > This ended up being a simple canreinvite situation... I had yes - and needed to be set to NO. Jerry
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