[asterisk-users] Failed to authenticate
Jerry Geis
jerry.geis at gmail.com
Wed Aug 11 08:10:54 CDT 2021
On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis <jerry.geis at gmail.com> wrote:
>
>
> On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis <jerry.geis at gmail.com> wrote:
>
>>
>>
>> On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis <jerry.geis at gmail.com> wrote:
>>
>>>
>>>
>>> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis <jerry.geis at gmail.com> wrote:
>>>
>>>>
>>>>
>>>> On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis <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 ?
Thanks
Jerry
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