asterisk users - Feb 2019 - Asterisk - can't hear other side. Or other side does not hear us

Antony,
It is correct. Noone connects to Asterisk box/server from outside.Callcentric
SIP trunk configured and Asterisk maintains connection to it itself. No special
ports opened, nothing. Connection happens from us to Callcentric and all calls
routed in from CallcentricI don't know exactly how it's doing it by it
works.
Again, keep in mind it is working for many years for most / 90+% of calls
#1. I don't think it will be possible to know :( Been 3 years.
#2-3. All callers call public phone number and they all come in to asterisk from
Callcentric context.When we call out - it goes out through Callcentric SIP
trunk.
When we dial internal each others extensions there is no NAT, trunk or anything
else and all works just fine...
Debugging with "tshark" should be done on Asterisk machine I asume? 
Thank you!

On Thursday 28 February 2019 at 00:26:17, Ivan Demkovitch wrote:
> Asterisk is NOT exposed to internet, noone connects to Asterisk> from
internet. We use Callcentric for VOIP trunk.
That's the point where you lost me.

Callcentric is out on the Internet.  How does it connect to your Asterisk 
server?
> External callers get in via Callcentric.
Right...
> 1. Outside caller calls us but can't hear us. I beleive they talked to
their> phone provider and it works now?
It would be good to know what got changed to make that work.
> 2. We have one caller where EVERY time they call - they can't hear us.
They> just say "ok, call us back". We call back and it works :)
So, they connect in via Callcentric too?  Just the same as Caller 1?
> 3. We have one caller where when we call them - they cannot> hear us,
but we can hear them. They called back - all works.
What is the difference between callers 1, 2 and 3, in terms of how they connect 
to your Asterisk server, or how you connect to them?
> I feel like we need to trace SIP protocol. How do I do that? I may get
on> of those callers to work with us on testing.
I would start with something like:

# tshark -i any -f "port 5060" -w "sip.debug.pcap"

and then afterwards look at the pcap file with tshark (tshark -r 
"sip.debug.pcap -V") or some SIP tool such as sngrep.


Antony.

 

 
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On Thursday 28 February 2019 at 17:40:28, Ivan Demkovitch wrote:
> Noone connects to Asterisk box/server from outside.  Callcentric SIP trunk
> configured and Asterisk maintains connection to it itself.
Okay, I didn't actually mean "does anyone connect *inbound* to your
Asterisk
server" - I was more asking about the connectivity between the Asterisk box
and (anything on) the Internet (eg: Callcentric), and whether there is NAT 
involved.
> No special ports opened, nothing. Connection happens from us
> to Callcentric and all calls routed in from CallcentricI don't know
> exactly how it's doing it by it works.
Does your Asterisk box have an RFC1918 address (ie: 10.0.0.0/8. 172.16.0.0/12 
or 192.168.0.0/16), or does it have a public IP address on its own interface?

If the address on your Asterisk server falls in the RFC1918 range, then you 
have NAT occurring on your router, and this is known to cause one-way (or 
sometimes no-way) audio from time to time.
> Debugging with "tshark" should be done on Asterisk machine I
asume?
Yes, or else on any router between the Asterisk machine and the other end of 
the affected calls (ie: Callcentric).  It's very probably simplest to do it
on
the Asterisk server, though.


Antony.

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Antony,
Ok, I see what you are saying. Yes, than NAT occuring on our router. Asterisk
server is on internal IP (192.168..)
# Now that I read what you say I think there might be 2 issues.
"Randomness" is one, but I am not even sure we have it (randomness).
All recent complains were from specific callers and we can replicate those.
#What am I looking for? Should it be "tshark" or there is means within
Asterisk application/configs to write those logs?I haven't used
"tshark" before. And more specifically, what am I looking for?
Thank you!
On Thursday 28 February 2019 at 17:40:28, Ivan Demkovitch wrote:
> Noone connects to Asterisk box/server from outside. Callcentric SIP
trunk> configured and Asterisk maintains connection to it itself.
Okay, I didn't actually mean "does anyone connect *inbound* to your
Asterisk
server" - I was more asking about the connectivity between the Asterisk box
and (anything on) the Internet (eg: Callcentric), and whether there is NAT 
involved.
> No special ports opened, nothing. Connection happens from us> to
Callcentric and all calls routed in from CallcentricI don't know> exactly
how it's doing it by it works.
Does your Asterisk box have an RFC1918 address (ie: 10.0.0.0/8. 172.16.0.0/12 
or 192.168.0.0/16), or does it have a public IP address on its own interface?

If the address on your Asterisk server falls in the RFC1918 range, then you 
have NAT occurring on your router, and this is known to cause one-way (or 
sometimes no-way) audio from time to time.
> Debugging with "tshark" should be done on Asterisk machine I
asume?
Yes, or else on any router between the Asterisk machine and the other end of 
the affected calls (ie: Callcentric).  It's very probably simplest to do it
on
the Asterisk server, though.


Antony.

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On Thursday 28 February 2019 at 18:00:54, Ivan Demkovitch wrote:
> Antony,
> Ok, I see what you are saying. Yes, than NAT occuring on our router.
> Asterisk server is on internal IP (192.168..)
> # Now that I read what you say I think there might be 2 issues.
"Randomness"
> is one, but I am not even sure we have it (randomness). All recent
complains
> were from specific callers and we can replicate those.
That is significant.
> #What am I looking for? Should it be "tshark" or there is means
within
> Asterisk application/configs to write those logs?
There is no way to get a full SIP trace out of Asterisk (that I know of).
> I haven't used "tshark" before. And more specifically, what
am I looking for?
tshark is just my tool of choice for capturing packets (and, sometimes, for 
analysing what they mean afterwards).  tcpdump is the other obvious choice for 
capturing the network traffic so you can then feed it into a protocol analyser.

I did suggest sngrep as being another useful way of analysing what's going
on;
this is a SIP-specific tool for showing you the flow of a conversation between 
two SIP devices (such as your Asterisk server and Callcentric's).

In terms of what you're looking for, I'd start by looking at the
exchange of
media negotiation - do both servers support at least one codes acceptable to 
the other?  And, are both servers telling the other to send the media to a 
valid public address?


Regards,


Antony.

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The barman asks him "Do you want a drink?"
Descartes says "I think not," and disappears.

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