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<p>Hello<br>
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<div class="moz-cite-prefix">Le 11/08/2021 à 15:10, Jerry Geis a
écrit :<br>
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cite="mid:CABr8-B79GNK5JyzO12DUWqdt-RGaBW30Bx4kUgDhs03b+SE+=A@mail.gmail.com">
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<div dir="ltr" class="gmail_attr">On Mon, Aug 9, 2021 at 11:05
AM Jerry Geis <<a href="mailto:jerry.geis@gmail.com"
moz-do-not-send="true">jerry.geis@gmail.com</a>> wrote:<br>
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<div dir="ltr" class="gmail_attr">On Mon, Aug 9, 2021 at
8:32 AM Jerry Geis <<a
href="mailto:jerry.geis@gmail.com" target="_blank"
moz-do-not-send="true">jerry.geis@gmail.com</a>>
wrote:<br>
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<div dir="ltr" class="gmail_attr">On Mon, Aug 9,
2021 at 8:11 AM Jerry Geis <<a
href="mailto:jerry.geis@gmail.com"
target="_blank" moz-do-not-send="true">jerry.geis@gmail.com</a>>
wrote:<br>
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<div dir="ltr" class="gmail_attr">On Mon,
Aug 9, 2021 at 7:57 AM Jerry Geis <<a
href="mailto:jerry.geis@gmail.com"
target="_blank" moz-do-not-send="true">jerry.geis@gmail.com</a>>
wrote:<br>
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<div dir="ltr" class="gmail_attr">On
Sun, Aug 8, 2021 at 3:18 PM Jerry
Geis <<a
href="mailto:jerry.geis@gmail.com"
target="_blank"
moz-do-not-send="true">jerry.geis@gmail.com</a>>
wrote:<br>
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<div dir="ltr">I am not using a SIP
trunk as I normally do.
<div><br>
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<div>I have an extensions 3382
setup that my server registers
to the other SIP system.</div>
<div>When the other system calls
3381 on my system I am getting
this error:</div>
<div><br>
</div>
<div>[Jul 27 10:08:50]
WARNING[89791][C-00000068]
chan_sip.c: username mismatch,
have <3381>, digest has
<8124><br>
[Jul 27 10:08:50]
NOTICE[89791][C-00000068]
chan_sip.c: Failed to
authenticate device "USCOL TEST"
<a class="moz-txt-link-rfc2396E" href="sip:XXXX@IP"><sip:XXXX@IP></a>;tag=1c1947164290 for INVITE, code = -2<br>
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<div><br>
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<div>How I allow this ? I want
to allow any SIP call to 3381.
</div>
<div>Using Astering 18.4.0</div>
<div><br>
</div>
<div>Thanks,</div>
<div><br>
</div>
<div>Jerry</div>
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<div><br>
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<div>Sure here it is:</div>
[general](+)<br>
register => 3382:XX@IP/3382<br>
<br>
; Description: Connection to PBX<br>
[3382]<br>
type=friend<br>
defaultname=3382<br>
defaultuser=3382<br>
secret=XX<br>
dtmfmode=RFC2833<br>
host=IP<br>
description=Connection to PBX<br>
context=incoming<br>
rtptimeout=60<br>
rtpholdtimeout=60<br>
rtpkeepalive=60<br>
callerid=3382<br>
qualify=no<br>
canreinvite=no<br>
nat=never<br>
disallow=all<br>
allow=ulaw<br>
allow=alaw<br>
allow=gsm</div>
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<div class="gmail_quote">Thanks </div>
<div class="gmail_quote">Jerry</div>
<div class="gmail_quote"><br>
</div>
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<div><br>
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<pre style="white-space:pre-wrap;color:rgb(0,0,0)">> What's the association between 3381 and 3382?
</pre>
3381 is the number they want to dial into my
asterisk. 3382 is the registered extension
to their system.</div>
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<div class="gmail_quote">Jerry</div>
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<div>><span style="color:rgb(0,0,0);white-space:pre-wrap">You register as 3382. That means that if someone on their system dials 3382, </span></div>
<div><span style="color:rgb(0,0,0);white-space:pre-wrap">>your Asterisk server gets the call.</span> </div>
<div><br>
</div>
<div><br>
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<div>I think at first I was only using 3381. That
was the extension I registered. There was no
3382. Something was going wrong there also.
(Might have been a similar error), </div>
<div>and I could not get that to work either.</div>
<div><br>
</div>
<div>Jerry</div>
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<div>Well my issue has changed now. I have dropped the
3382. Changed back to 3381. So I am registering 3381
to the other server.</div>
<div>The other server is 10.35.229.5. My IP is
10.35.229.11. </div>
<div>I have two network cards.</div>
<div><br>
</div>
<div>10.35.229.11 is Eth0</div>
<div>192.168.1.60 is Eth1</div>
<div><br>
</div>
<div>route looks OK</div>
<div>route -n<br>
Kernel IP routing table<br>
Destination Gateway Genmask Flags
Metric Ref Use Iface<br>
0.0.0.0 192.168.1.1 0.0.0.0 UG
0 0 0 eth1<br>
10.35.229.0 0.0.0.0 255.255.255.0 U
0 0 0 eth0<br>
169.254.0.0 0.0.0.0 255.255.0.0 U
1002 0 0 eth0<br>
169.254.0.0 0.0.0.0 255.255.0.0 U
1003 0 0 eth1<br>
192.168.1.0 0.0.0.0 255.255.255.0 U
0 0 0 eth1<br>
</div>
<div><br>
</div>
<div>The issue is that the call comes in but the user
hears no audio.</div>
<div>There is any crazy networking going on - why would
the user not hear audio ?</div>
<div>Thanks</div>
<div><br>
</div>
<div>Jerry </div>
</div>
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<div>Hello All,</div>
<div><br>
</div>
<div>I got more information about the "no audio".</div>
<div><br>
</div>
<div>The incoming call is from 10.37.229.5 - I have two
network cards in the box.</div>
<div>10.35.229.11 eth0 </div>
<div>192.168.1.60 eth1 </div>
<div><br>
</div>
<div>When I noticed the incoming address was 10.37.229.5 I
thought the audio packets are sending out the default route
of eth1.</div>
<div>SO I tried to add a route:</div>
route -n<br>
Kernel IP routing table<br>
Destination Gateway Genmask Flags Metric
Ref Use Iface<br>
0.0.0.0 192.168.1.1 0.0.0.0 UG 0 0
0 eth1<br>
10.35.229.0 0.0.0.0 255.255.255.0 U 0 0
0 eth0<br>
10.37.229.0 0.0.0.0 255.255.255.0 U 0 0
0 eth0<br>
169.254.0.0 0.0.0.0 255.255.0.0 U 1002 0
0 eth0<br>
169.254.0.0 0.0.0.0 255.255.0.0 U 1003 0
0 eth1<br>
<div>192.168.1.0 0.0.0.0 255.255.255.0 U 0
0 0 eth1</div>
<div><br>
</div>
<div>But I am still not getting audio.</div>
<div><br>
</div>
<div>Anything else I might try ?</div>
</div>
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<p>Check if your networks in localnet are correctly defined.<br>
</p>
<pre class="moz-signature" cols="72">--
Daniel</pre>
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