Hello,
it triggered again. Even disabling RTSp ALG did not help. My fantasy
ends here. It agains seems to be reboot triggered on asterisk side.
Not every one. But there was surely one before it was last working.
Reboot of the router on the phone side fixes the problem. Any other
suggestions?
This is where you use sngrep or tcpdump to look at whats actually
happening on the asterisk box. sngrep is focussed on sip dialogs and is
probably easier than tcpdump when you are just interested in sip
If you use sngrep on the asterisk server sip port you will see the SIP
packet flows for registration and call setups. You can check the
addresses given out for rtp to respond to and the codecs. Is an address
incorrect? Is a code incorrect? You will see in the session description
protocol what codecs the client is requesting and what the replies are
asterisk works well around the world with many nat scenarios so I
imagine its either config or firewall. A firewall with ALGs is often
problematic but your log suggests a lack of negotiation of agreed
codecs.
Good luck, you will learn some interesting things.
Thanks
Marek
2021-07-26 9:31 GMT+02:00, Marek Greško <mgres...@gmail.com>:
I currently disabled also RTSP ALG and rebooted the router. Fixed for
now. I do not know for how long.
Marek
2021-07-26 8:54 GMT+02:00, Marek Greško <mgres...@gmail.com>:
Hmm, back to original problem. My happines was premature. Today one of
the phones have no audio again. I see packets from lan segment again.
I double checked the router configuration. SIP ALG is disabled. There
are also another ALGs present:
NAT ALG
RTSP ALG
PPTP ALG
IPSEC ALG
Which of them are neede to be disabled?
As of my observations these problems are triggered by reboots on
asterisk side. How could this be related? (I may be wrong.)
Thanks
Marek
2021-07-23 14:54 GMT+02:00, Marek Greško <mgres...@gmail.com>:
I achieved a partial success adding --use-compact-form option.
Marek
2021-07-23 13:47 GMT+02:00, Marek Greško <mgres...@gmail.com>:
Hello,
your suggestion to turn off SIP ALG on provider's router was probably
correct. no problem until now. Thank you very much.
I just found out another issue. I had a pjsue client in that network
which called specific number when turned on. It was working perfectly
with the old provider with working SIP ALG. But now with this provider
and SIP ALG disabled I am not able to make the call using pjsua
client.
My pjsua config looks like this:
--id sip:ext@asterisk.domain
--registrar sip:asterisk.domain
--proxy sip:asterisk.domain
--outbound sip:asterisk.domain
--realm *
--username username
--password password
--null-audio
--no-tcp
--max-calls=1
--no-vad
The pjsua client successfully registers but is unable to call.
I see the following:
IP address change detected for account 1
(localip:5060-->nattedip:newport). Updating registration (using method
4)
Temporary failure in sending Request msg INVITE/cseq=...., will try
next server: Unsupported transport (PJSIP_EUNSUPTRANSPORT)
What could be the problem? How can I convince pjsue to work correctly
behind nat?
Thanks
Marek
2021-07-10 11:08 GMT+02:00, Marek Greško <mgres...@gmail.com>:
Hello,
I just disabled. Currently it is working. I shloud give it some time
to confirm the problem has gone. Maybe one month would be enough to
confirm.
Thanks
Marek
2021-07-09 20:11 GMT+02:00, Abdenasser Ghomri
<ghomri.nas...@gmail.com>:
Yes just disable the SIP ALG and see if it helps, Thanks.
Best Regards,
On Fri, Jul 9, 2021, 09:10 Antony Stone <
antony.st...@asterisk.open.source.it> wrote:
On Friday 09 July 2021 at 08:47:46, Marek Greško wrote:
> Hello,
>
> yes SIP ALG are anbled on the router. Should I disable?
In my opinion, always.
Antony.
--
I don't know, maybe if we all waited then cosmic rays would write
all
our
software for us. Of course it might take a while.
- Ron Minnich, Los Alamos National Laboratory
Please reply to
the
list;
please
*don't*
CC
me.
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