Hi,
I made some interesting observations regarding this.
Remind the following scenario:
asterisk registers number A towards provider A (sipgate.de)
asterisk also registers another number towards provider B (tel.t-online.de)
I make a test call from a remote location, which is registered as well
ok, now it is getting weird...
actually i don't see any firewall issues, but i am not able to get a
call from outside to my sipgate account. in asterisk nothing is visible,
core set verbose is activated.
sngrep (on my asterisk server) shows me indeed the invite from sipgate!?
What I see via
Have you tried setting keepalive(20 seconds) on your sip.conf and on your
phones ?
From: Andre Gronwald
To: asterisk-users@lists.digium.com
Sent: Saturday, October 15, 2016 9:17 AM
Subject: Re: [asterisk-users] Registered successfully, but after a minute
ok, solved the firewall issue.
A first test call worked fine. Another one now still gets disconnected
after 32s.
But in FW there are no blocked packets anymore?!
And I don't understand why the registration to the same IP and same Port
is working, but not later transmission of further SIP
Hi,
I don’t see any SIP ACK’s in your trace.
Is the SIP 200 OK reaching the originating caller, or being blocked on
the way through?
Asterisk will tear down the call after ~30secs of audio playing in both
directions if it doesn't receive the SIP ACK.
Regards,
Ian
On 15/10/2016 12:05,
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT,
Thanks Jonathan for your support.
I would like to avoid TLS at the moment (in general I am a fan of
secured communication!) because the other provider is not supporting
TLS. And sipgate is just used for testing.
However I can see the following which is quite interesting:
[2016-10-15
Hmmm, sorry, I can't think of anything except... why do you need the
STUN server? And are you sure that all the ports in your router
definitely match the ones Asterisk thinks it's using?
Then there is always the SIP-ALG problem with some routers, which some
people have been able to overcome by
ping times are fine as well:
[root@freepbx asterisk]# ping sipgate.de
PING sipgate.de (217.10.79.9) 56(84) bytes of data.
64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms
64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms
64 bytes from sipgate.de
All other things aside, this stands out immediately:
RTT: 434.393 msec
That's almost half a second round trip for a packet. I'm amazed
anything works at all. For SIP connections, mine are usually about
26ms max, anything above about 35 is bad. Looks like a serious config
issue.
Try pinging and
Very interesting: I have another provider configured, that was not
reachable as well. I disabled the STUN-server (external STUN server),
and now the second registration works fine, BUT with the same "error"
messages (unreachable etc) as the other provider. But in contrast the
number is always
Hi all,
I have an issue with asterisk 13 and pjsip. I guess it is somehow
Firewall related, but I'm unsure.
A registration to Sipgate is established successfully:
==
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