Re: [asterisk-users] Failed to authenticate

2021-08-11 Thread Administrator 

Hello

Le 11/08/2021 à 15:10, Jerry Geis a écrit :



On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis > wrote:




On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis mailto:jerry.g...@gmail.com>> wrote:



On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis
mailto:jerry.g...@gmail.com>> wrote:



On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis
mailto:jerry.g...@gmail.com>> wrote:



On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis
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-0068]
chan_sip.c: username mismatch, have <3381>, digest
has <8124>
[Jul 27 10:08:50] NOTICE[89791][C-0068]
chan_sip.c: Failed to authenticate device "USCOL
TEST" ;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

-- 
_
--

Re: [asterisk-users] Failed to authenticate

2021-08-11 Thread Jerry Geis 
On Mon, Aug 9, 2021 at 11:05 AM Jerry Geis  wrote:

>
>
> On Mon, Aug 9, 2021 at 8:32 AM Jerry Geis  wrote:
>
>>
>>
>> On Mon, Aug 9, 2021 at 8:11 AM Jerry Geis  wrote:
>>
>>>
>>>
>>> On Mon, Aug 9, 2021 at 7:57 AM Jerry Geis  wrote:
>>>


 On Sun, Aug 8, 2021 at 3:18 PM Jerry Geis  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-0068] chan_sip.c: username
> mismatch, have <3381>, digest has <8124>
> [Jul 27 10:08:50] NOTICE[89791][C-0068] chan_sip.c: Failed to
> authenticate device "USCOL TEST" ;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 RefUse
> Iface
> 0.0.0.0 192.168.1.1 0.0.0.0 UG0  00
> eth1
> 10.35.229.0 0.0.0.0 255.255.255.0   U 0  00
> eth0
> 169.254.0.0 0.0.0.0 255.255.0.0 U 1002   00
> eth0
> 169.254.0.0 0.0.0.0 255.255.0.0 U 1003   00
> eth1
> 192.168.1.0 0.0.0.0 255.255.255.0   U 0  00
> 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 RefUse
Iface
0.0.0.0 192.168.1.1 0.0.0.0 UG0  00 eth1
10.35.229.0 0.0.0.0 255.255.255.0   U 0  00 eth0
10.37.229.0 0.0.0.0 255.255.255.0   U 0  00 eth0
169.254.0.0 0.0.0.0 255.255.0.0 U 1002   00 eth0
169.254.0.0 0.0.0.0 255.255.0.0 U 1003   00 eth1
192.168.1.0 0.0.0.0 255.255.255.0   U 0  00 eth1

But I am still not getting audio.

Anything else I might try ?

Thanks

Jerry
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