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