[asterisk-users] Failed to authenticate

[Administrator]

Hello

Le 11/08/2021 à 15:10, Jerry Geis a écrit :
>
>
> On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis <jerry.geis at gmail.com 
> <mailto:jerry.geis at gmail.com>> wrote:
>
>
>
>     On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis <jerry.geis at gmail.com
>     <mailto:jerry.geis at gmail.com>> wrote:
>
>
>
>         On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis
>         <jerry.geis at gmail.com <mailto:jerry.geis at gmail.com>> wrote:
>
>
>
>             On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis
>             <jerry.geis at gmail.com <mailto:jerry.geis at gmail.com>> wrote:
>
>
>
>                 On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis
>                 <jerry.geis at gmail.com <mailto:jerry.geis at 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 at 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 at 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20210811/e23f7e44/attachment.html>