Andre Gronwald
2016-Oct-15 13:17 UTC
[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore
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 packets? that doesn't sound logical to me. What do you think? regards, andre
Edilson Amaral
2016-Oct-15 13:37 UTC
[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore
Have you tried setting?keepalive(20 seconds) on your sip.conf and on your phones
?
From: Andre Gronwald <andregronwald78 at gmail.com>
To: asterisk-users at lists.digium.com
Sent: Saturday, October 15, 2016 9:17 AM
Subject: Re: [asterisk-users] Registered successfully, but after a minute or so
no SIP messages anymore
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 packets? that
doesn't sound logical to me. What do you think?
regards,
andre
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Andre Gronwald
2016-Oct-15 13:39 UTC
[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore
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 sngrep is the following options-flow: 192.168.3.50:55060 -> 217.10.79.9:5060 217.10.79.9:5060 -> 192.168.3.50:48757 (200 OK) shouldn't sipgate answer on the same port that the communication initiated??? in this case 55060? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20161015/79c208f3/attachment.html>
Andre Gronwald
2016-Oct-17 16:41 UTC
[asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore
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
towards provider B.
What happens on the asterisk?
have a look below.
It seems that the registration to number B of provider B, which sends
the incoming call to asterisk, brings asterisk out of order, because the
ACKs are sent to the wrong port.
after some time the ACK of provider B reaches asterisk on the right port
(55060), but that is maybe by accident. when the trunk to provider B is
disabled it works fine immediately.
That explains, why calls to sipgate.de are cancelled after about 30s.
But can someone explain that? Is it an issue of provider B (which I
can't believe, it is one of the biggest here in germany)?
217.10.79.9:5060 10.17.46.99:55060
217.0.23.36:5060 10.17.46.99:65319 10.17.46.99:505
???????????????????? ????????????????????
???????????????????? ???????????????????? ?????????????????
? ? INVITE (SDP) ?
? ? ?
? 18:26:47.695820 ? ??????????????????????????>
? ? ? ?
? +0.001354 ? 100 Trying ? ?
? ?
? 18:26:47.697174 ? <??????????????????????????
? ? ? ?
? +0.024128 ? 200 OK (SDP) ?
? ? ?
? 18:26:47.721302 ? <??????????????????????????
? ? ? ?
? +0.418916 ? INVITE (SDP) ?
? ? ?
? 18:26:48.140218 ? ????????????????????????>>>
? ? ? ?
? +0.000464 ? 200 OK (SDP) ?
? ? ?
? 18:26:48.140682 ? <<<????????????????????????
? ? ? ?
? +0.047604 ? ? ? ACK
? ?
? 18:26:48.188286 ? ? ?
??????????????????????????> ? ?
? +0.033199 ? 200 OK (SDP) ?
? ? ?
? 18:26:48.221485 ? <<<????????????????????????
? ? ? ?
? +0.413234 ? ? ? ACK
? ?
? 18:26:48.634719 ? ? ?
??????????????????????????> ? ?
? +0.073871 ? ? ? ACK
? ?
? 18:26:48.708590 ? ? ?
??????????????????????????> ? ?
? +1.513110 ? 200 OK (SDP) ?
? ? ?
? 18:26:50.221700 ? <<<????????????????????????
? ? ? ?
? +0.478899 ? ? ? ACK
? ?
? 18:26:50.700599 ? ? ?
??????????????????????????> ? ?
? +3.520824 ? 200 OK (SDP) ?
? ? ?
? 18:26:54.221423 ? <<<????????????????????????
? ? ? ?
? +0.524939 ? ? ? ACK
? ?
? 18:26:54.746362 ? ? ?
??????????????????????????> ? ?
? +3.475120 ? 200 OK (SDP) ?
? ? ?
? 18:26:58.221482 ? <<<????????????????????????
? ? ? ?
? +0.482321 ? ? ? ACK
? ?
? 18:26:58.703803 ? ? ?
??????????????????????????> ? ?
? +3.519206 ? 200 OK (SDP) ?
? ? ?
? 18:27:02.223009 ? <<<????????????????????????
? ? ? ?
? +0.481228 ? ? ? ACK
? ?
? 18:27:02.704237 ? ? ?
??????????????????????????> ? ?
? +3.518524 ? 200 OK (SDP) ?
? ? ?
? 18:27:06.222761 ? <<<????????????????????????
? ? ? ?
? +0.483722 ? ? ? ACK
? ?
? 18:27:06.706483 ? ? ?
??????????????????????????> ? ?
? +3.516816 ? 200 OK (SDP) ?
? ? ?
? 18:27:10.223299 ? <<<????????????????????????
? ? ? ?
? +0.484398 ? ? ACK ?
? ?
? 18:27:10.707697 ? ?
<?????????????????????????? ? ? ?
? +59.790625 ? ? ?
BYE ?
? 18:28:10.498322 ? ? ?
????????????????????????????????????????????????????????> ?
Am 15.10.2016 um 15:39 schrieb Andre Gronwald:> 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 sngrep is the following options-flow:
> 192.168.3.50:55060 -> 217.10.79.9:5060
> 217.10.79.9:5060 -> 192.168.3.50:48757 (200 OK)
>
> shouldn't sipgate answer on the same port that the communication
> initiated??? in this case 55060?
>
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