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

[Jerry Geis]

On Fri, Feb 4, 2022 at 12:42 PM Jerry Geis <jerry.geis at gmail.com> wrote:

>
>
> 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
>


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