[asterisk-users] Failed to authenticate

[Eric Wieling]

You could switch to PJSIP and avoid most of this silliness.

I love Asterisk, but the peer/user/friend model in chan_sip is simply 
terrible.

PJSIP is different so there is a learning curve, of course.

On 8/9/21 11:05 AM, Jerry Geis 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
> 

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