[asterisk-users] Voice "broken" during calls
lucabert at lucabert.de
Tue Jun 16 04:23:27 CDT 2020
Am 16.06.2020 10:48, schrieb Antony Stone:
> On Tuesday 16 June 2020 at 08:18:51, Luca Bertoncello wrote:
>> > sudo tcpdump -i eth0 -s 0 -w /tmp/test0.pcap &
>> > sudo tcpdump -i eth1 -s 0 -w /tmp/test1.pcap &
>> eth0 is my DSL interface and eth1 my phone interface?
> Well, one is internal (phone) and the other is external (DT), doesn't
> which way round.
This was what I meant...
>> tcpdump -i dsl0 -s 0 -w /tmp/test0.pcap host tel.t-online.de &
>> tcpdump -i phone0 -s 0 -w /tmp/test1.pcap host 192.168.200.xx (IP of
>> phone) &
> Looks like you name your Banana interfaces very similarly to mine :)
I think, we are not alone... :D
> However, I would be careful with that first one, containing "host
> online.de". I don't use DT, so I can't be sure, but I guess this is
> the SIP
> server to which you register with the account credentials...
> It *may not* be the same machine as handles the RTP packets - that is
> negotiated separately between Asterisk (or the Thomson, when it's
> directly to DT) as part of the SIP INVITE / Acknowledge.
> So, you *could* find that you capture all of the SIP traffic and none
> of the RTP
> traffic. On the other hand, you might get everything.
> You can be pretty sure it's worked if you do the above and then find
> that the
> two packet capture files are approximately the same size. If the DT
> one is
> significantly smaller (by which I mean a factor of at least ten
> different), then
> omit the "host" parameter on that capture and try again...
OK, I'll check it...
(lucabert at lucabert.de)
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