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

Hello,
This is not technical post, just looking for suggestions on what to check.I have
asterisk for long time, no updates, just maintain OS updates.
I use SPA504G phones
Very rarely and randomly when we pickup a phone - other side does not hear us.
Call them back and all works.
Now I have couple people I'm talking to and it seems like very call like
this. Someone can't hear someone.
Don't know where to start to troubleshoot and what to look for.
Thanks!
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On Wednesday 27 February 2019 at 23:10:33, Ivan Demkovitch wrote:
> Hello,
> This is not technical post,
Hm, no?
> just looking for suggestions on what to check.I have asterisk for long
time,
Which version?
> no updates, just maintain OS updates.  I use SPA504G phones.
Tell us about your network - where is Asterisk (inside your network, 
externally hosted on public IP address, other), where are your phones (inside 
one network (maybe the same as Asterisk is on), randomly distributed around 
the Internet, other), how do external callers manage to contact you?
> Very rarely and randomly when we pickup a phone - other side does not hear
> us. Call them back and all works. Now I have couple people I'm talking
to
> and it seems like very call like this. Someone can't hear someone.
Don't
> know where to start to troubleshoot and what to look for.
Short answer: NAT

Longer answer: Check the type of firewall / router / NAT device you have 
between Asterisk and the phones (most likely at the telephone end) and see 
whether it offers "SIP ALG" (Application Layer Gateway) - if it does,
turn it
*off*.

Also, check the sip.conf definitions you have for the phones which are affected 
by this, and make sure you have NAT set to one of yes, force_rport or comedia 
(you may hav eto experiment to see which works best in your environment).

Check https://www.voip-info.org/asterisk-sip-nat/ for some guidance.

https://www.voip-info.org/asterisk-sip-nat-solutions/ may also give you some 
further clues.

On the other hand, note that https://www.voip-info.org/asterisk-config-sipconf/ 
is woefully outdated (at least as far as NAT is concerned).


If none of that helps, I suggest doing a SIP packet trace at the server (and 
at the phone end if you can) and see what addresses are being passed between 
the two for RTP.  That should tell you why one end can't contact the other.


Regards,


Antony.

-- 
The gravitational attraction exerted by a single doctor at a distance of 6 
inches is roughly twice that of Jupiter at its closest point to the Earth.

                                                   Please reply to the list;
                                                         please *don't* CC
me.

Antony, thanks for response!
It wasn't technical, now it's getting there :)
1. It's asterisk 13.1-cert12. Network. I actually tried multiple things
initially but now it's plain vanilla. No NAT. I have Asterisk on our
network. All of our phones is IN our network as well, same subnet. All internal.
Asterisk is NOT exposed to internet, noone connects to Asterisk from internet.
We use Callcentric for VOIP trunk. External callers get in via Callcentric.
We have Mikrotik router and SIP handlers were disabled from beginning
I gathered more info now about 3 issues we seen1. Outside caller calls us but
can't hear us. I beleive they talked to their phone provider and it works
now?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 :)3. We
have one caller where when we call them - they cannot hear us, but we can hear
them. They called back - all works.
So, as you see we don't have NAT stuff
[general]
dtmfmode=rfc2833
context=unauthenticated
allowguest=no
srvlookup=no
udpbindaddr=0.0.0.0
tcpenable=no
callcounter=yes                 ; This is to enable device state for queues
nat=no
session-timers=refuse
type=peer
context=internal
host=dynamic
disallow=all
allow=ulaw
allow=alaw

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.

Thanks!
> Hello,> This is not technical post,
Hm, no?
> just looking for suggestions on what to check.I have asterisk for long
time,
Which version?
> no updates, just maintain OS updates. I use SPA504G phones.
Tell us about your network - where is Asterisk (inside your network, 
externally hosted on public IP address, other), where are your phones (inside 
one network (maybe the same as Asterisk is on), randomly distributed around 
the Internet, other), how do external callers manage to contact you?
> Very rarely and randomly when we pickup a phone - other side does not
hear> us. Call them back and all works. Now I have couple people I'm
talking to> and it seems like very call like this. Someone can't hear
someone. Don't> know where to start to troubleshoot and what to look for.
Short answer: NAT

Longer answer: Check the type of firewall / router / NAT device you have 
between Asterisk and the phones (most likely at the telephone end) and see 
whether it offers "SIP ALG" (Application Layer Gateway) - if it does,
turn it
*off*.

Also, check the sip.conf definitions you have for the phones which are affected 
by this, and make sure you have NAT set to one of yes, force_rport or comedia 
(you may hav eto experiment to see which works best in your environment).

Check https://www.voip-info.org/asterisk-sip-nat/ for some guidance.

https://www.voip-info.org/asterisk-sip-nat-solutions/ may also give you some 
further clues.

On the other hand, note that https://www.voip-info.org/asterisk-config-sipconf/ 
is woefully outdated (at least as far as NAT is concerned).


If none of that helps, I suggest doing a SIP packet trace at the server (and 
at the phone end if you can) and see what addresses are being passed between 
the two for RTP.  That should tell you why one end can't contact the other.


Regards,


Antony.

 
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This is usually a symptom of something on the call path mishandling the session
setup.  Check routers/firewall/SIP proxy, etc.  Likely a firmware bug is causing
it to use the wrong IP address and passing that to the other end.

 

Even if you disabled these devices, REMOVE them from the call path (or replace)
for testing.  Add them back one at a time to confirm source of problem.

 

Sue

 

From: asterisk-users [mailto:asterisk-users-bounces at lists.digium.com] On
Behalf Of Ivan Demkovitch
Sent: Wednesday, February 27, 2019 5:11 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] Asterisk - can't hear other side. Or other side
does not hear us

 

Hello,

 

This is not technical post, just looking for suggestions on what to check.

I have asterisk for long time, no updates, just maintain OS updates.

 

I use SPA504G phones

 

Very rarely and randomly when we pickup a phone - other side does not hear us.
Call them back and all works.

 

Now I have couple people I'm talking to and it seems like very call like
this. Someone can't hear someone.

 

Don't know where to start to troubleshoot and what to look for.

 

Thanks!

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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.

-- 
"It wouldn't be a good idea to talk about him behind his back in front
of
him."

 - murble

                                                   Please reply to the list;
                                                         please *don't* CC
me.

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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