[asterisk-users] Odd issue CentOS 7 vs Ubuntu 20.04

[Jerry Geis]

On Wed, Feb 2, 2022 at 1:06 PM Jerry Geis <jerry.geis at gmail.com> wrote:

>
>
> On Wed, Feb 2, 2022 at 10:44 AM Jerry Geis <jerry.geis at gmail.com> wrote:
>
>>
>>
>> On Wed, Feb 2, 2022 at 9:26 AM Jerry Geis <jerry.geis at 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
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