subject:"\[Asterisk\-Users\] Failed to authenticate"

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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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Julian Beach

Hello Antony,

Monday, August 9, 2021, 4:14:11 PM, you wrote:

> You want to look for firewall rules which will allow UDP in both directions 
> on 
> ports 1 - 3 (typically, may vary a bit, but something like that), or 
> alternatively, look for any rules which would block this, and remove them.

This. Check that the port range in rdp.conf matches that in your firewall UDP 
settings. Even a slight mismatch in the ranges can result in a surprising 
number of calls with one-way audio which seem to happen in clusters.  

-- 
Best regards,
 Julianmailto:jb_s...@trink.co.uk

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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread 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 > 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

--
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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Antony Stone

On Monday 09 August 2021 at 17:05:42, Jerry Geis wrote:

> Well my issue has changed now.  I have dropped the 3382. Changed back to
> 3381.   So I am registering 3381 to the other server.

That makes more sense to me, at least.

> 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

I think eth1 and your routing table are not relevant to this.

> 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?

Commonly, because of firewalling and/or NAT.

Given that your client 10.35.229.11/24 and the server 10.35.229.5/24 are both 
on the same subnet, it's not going to be a NAT problem, so I would look at the 
firewall rules, both on your machine and the one you are connecting to.

(Please do tell us if the client and server are not connected directly through 
a switch as I have assumed here, and there's possibly something more 
complicated going on.)

You want to look for firewall rules which will allow UDP in both directions on 
ports 1 - 3 (typically, may vary a bit, but something like that), or 
alternatively, look for any rules which would block this, and remove them.

If that doesn't appear to be the problem, do a packet capture of your SIP 
traffic and look for the Invite and the reply, each with the SDP payloads, to 
find out what IP addresses and port numbers the client and server are 
advertising to each other.

The only other thing I can think of right now is codec compatibility.

Antony.

-- 
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specific.

   Please reply to the list;
 please *don't* CC me.

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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Jerry Geis

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
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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Jerry Geis

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
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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Antony Stone

On Monday 09 August 2021 at 14:11:18, Jerry Geis wrote:

> > 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.

Sorry - I'm confused by that.

You register as 3382.  That means that if someone on their system dials 3382, 
your Asterisk server gets the call.

I assume extension 8124 was placing the call in the example you gave:

>> [Jul 27 10:08:50] WARNING[89791][C-0068] chan_sip.c: username
>> mismatch, have <3381>, digest has <8124>

so I still don't understand where 3381 comes in.

Antony.

-- 
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 please *don't* CC me.

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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Jerry Geis

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



>
>
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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Antony Stone

On Sunday 08 August 2021 at 21:18:26, Jerry Geis wrote:

> 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:

What's the association between 3381 and 3382?


Antony.

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Re: [asterisk-users] Failed to authenticate

2021-08-09 Thread Jerry Geis

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
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Re: [asterisk-users] Failed to authenticate

2021-08-08 Thread Antony Stone

On Sunday 08 August 2021 at 21:18:26, Jerry Geis wrote:

> [Jul 27 10:08:50] WARNING[89791][C-0068] chan_sip.c: username mismatch,
> have <3381>, digest has <8124>

Show us the part of sip.conf which you use to register for this account.

Antony.

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[asterisk-users] Failed to authenticate

2021-08-08 Thread Jerry Geis

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
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Re: [asterisk-users] Failed to authenticate device message

2020-07-22 Thread Administrator


Hi Jerry

Le 22/07/2020 à 14:54, Jerry Geis a écrit :

I am getting this message:
Failed to authenticate device ;tag=149853321 for 
INVITE, code = -1


but it does not report the "connecting" address. Who is failing 
connecting ?
I either need to block someone or fix something - I'm thinking block - 
but I dont know who.

How do I found out the connecting IP?

Jerry


You should get it with recvip

exten = i,1,Verbose(Incoming ANONYMOUS SIP call from ${CALLERID(name)} 
${CALLERID(num)} SRC IP ${CHANNEL(recvip)})


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Re: [asterisk-users] Failed to authenticate device message

2020-07-22 Thread Telium Technical Support

You didn’t post the Asterisk version, but if this is an OLD asterisk version 
then the source IP may be missing from messages/logs.

 

If you have low traffic in general then using something like Wireshark may help 
you examine any suspicious SIP packet on the PBX.  For higher volumes it’s like 
drinking from a fire hydrant, so not suitable.

 

If this is a small PBX, have a look at the SecAst product 
(https://teium.io/secast).  It’s free for small installations.  It’s an 
Asterisk security product that monitors network traffic at a the adapter level 
so it can sniff the source.  It also talks to Asterisk through the AMI so it 
can get more details of the connection/session that way.  If this is for a 
larger PBX then you would have to move the discussion to the biz list for more 
info on SecAst.  (Or email me off list)

 

 

From: asterisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf 
Of Jerry Geis
Sent: Wednesday, July 22, 2020 11:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Subject: Re: [asterisk-users] Failed to authenticate device message

 

>Did you check your security log?
 
>There is usually a wealth of info there about who, what, where when and why
 
I also checked /var/log/asterisk/messages and it just has the same line. 
Nothing additional.
 
Jerry
 
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Re: [asterisk-users] Failed to authenticate device message

2020-07-22 Thread Jerry Geis

>Did you check your security log?

>There is usually a wealth of info there about who, what, where when and why


I also checked /var/log/asterisk/messages and it just has the same
line. Nothing additional.


Jerry
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Re: [asterisk-users] Failed to authenticate device message

2020-07-22 Thread Andrew Yager

Did you check your security log?

There is usually a wealth of info there about who, what, where when and why.

Andrew


On Wed, 22 Jul 2020 at 11:22 pm, Jerry Geis  wrote:

> >exten = i,1,Verbose(Incoming ANONYMOUS SIP call from ${CALLERID(name)}
> >${CALLERID(num)} SRC IP ${CHANNEL(recvip)})
>
>
> Thanks - its not an incoming call - its just a log on the CLI
> There is nothing before it and nothing after - no incoming call.
>
> Jerry
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Re: [asterisk-users] Failed to authenticate device message

2020-07-22 Thread Jerry Geis

>exten = i,1,Verbose(Incoming ANONYMOUS SIP call from ${CALLERID(name)}
>${CALLERID(num)} SRC IP ${CHANNEL(recvip)})


Thanks - its not an incoming call - its just a log on the CLI
There is nothing before it and nothing after - no incoming call.

Jerry
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[asterisk-users] Failed to authenticate device message

2020-07-22 Thread Jerry Geis

I am getting this message:
Failed to authenticate device ;tag=149853321 for INVITE,
code = -1

but it does not report the "connecting" address. Who is failing connecting ?
I either need to block someone or fix something - I'm thinking block - but
I dont know who.
How do I found out the connecting IP?

Jerry
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Re: [asterisk-users] Failed to authenticate device 100

2015-12-03 Thread Motty


Thanks M,
I have security enable,
; output security messages to the file named "Security"
security => security

I see the file created in /var/log/asterisk/security but is empty, and 
in /var/log/asterisk/messages I see the following:
[2015-12-03 06:52:32] NOTICE[19949] chan_sip.c: Failed to authenticate 
device 100<sip:100@X.X.X.X>;tag=a121ab55


X.X.X.X is the IP of my Server, I don't know who is the attacker IP 
unless I monitor for the server using the following command:

tcpdump -lni eth0 -f "udp port 5060"

Please advise.
Thanks,
Motty

On 12/02/2015 01:53 PM, Telium Technical Support wrote:


The details of the source IP are available in the asterisk security 
log (if you have that enabled) – but that particular attack hides its 
address from the messages file.


It’s essential that you secure your PBX; there are options ranging 
from free to commercial.  Have a look at:


http://www.voip-info.org/wiki/view/Asterisk+security

It’s easy to get a $20,000 phone bill, so take securing your PBX 
seriously.


-M-

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Motty

*Sent:* Wednesday, December 02, 2015 1:12 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion; 
motty.c...@gmail.com

*Subject:* [asterisk-users] Failed to authenticate device 100

Hello, I continued to see this errors in the logs:

[2015-12-02 10:05:57] NOTICE[19949]: chan_sip.c:23277 
handle_request_invite: Failed to authenticate device 
100<sip:1...@xx.xx.xx.xx> <mailto:sip:1...@xx.xx.xx.xx>;tag=10cdeaf7


how do I guard against this kinds of attacks? Also, to get the IP 
address from where this attack come from I use the following command 
"tcpdump -lni eth0 -f "udp port 5060" is there an easy way to get the 
attacker's IP?


Thanks,
Motty




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[asterisk-users] Failed to authenticate device 100

2015-12-02 Thread Motty


Hello, I continued to see this errors in the logs:

[2015-12-02 10:05:57] NOTICE[19949]: chan_sip.c:23277 
handle_request_invite: Failed to authenticate device 
100;tag=10cdeaf7


how do I guard against this kinds of attacks? Also, to get the IP 
address from where this attack come from I use the following command 
"tcpdump -lni eth0 -f "udp port 5060" is there an easy way to get the 
attacker's IP?


Thanks,
Motty
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Re: [asterisk-users] Failed to authenticate device 100

2015-12-02 Thread Telium Technical Support

The details of the source IP are available in the asterisk security log (if you 
have that enabled) – but that particular attack hides its address from the 
messages file.

 

It’s essential that you secure your PBX; there are options ranging from free to 
commercial.  Have a look at:

 

http://www.voip-info.org/wiki/view/Asterisk+security

 

It’s easy to get a $20,000 phone bill, so take securing your PBX seriously.

 

-M-

 

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motty
Sent: Wednesday, December 02, 2015 1:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; 
motty.c...@gmail.com
Subject: [asterisk-users] Failed to authenticate device 100

 

Hello, I continued to see this errors in the logs: 

[2015-12-02 10:05:57] NOTICE[19949]: chan_sip.c:23277 handle_request_invite: 
Failed to authenticate device 100 <mailto:sip:1...@xx.xx.xx.xx> 
<sip:1...@xx.xx.xx.xx>;tag=10cdeaf7

how do I guard against this kinds of attacks? Also, to get the IP address from 
where this attack come from I use the following command "tcpdump -lni eth0 -f 
"udp port 5060" is there an easy way to get the attacker's IP? 

Thanks, 
Motty

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[asterisk-users] Failed to authenticate device - who?

2014-12-10 Thread D'Arcy J.M. Cain

I have a bunch of these in my logs:

[Dec  9 08:21:21] NOTICE[-1][C-0285] chan_sip.c: Failed to
authenticate device
einsteinsip:einstein@98.158.139.74;tag=65696e737465696e0131323530333532333739

The problem is that I already know my own IP address.  How do I
determine the address of the host trying to hack my switch?

Cheers.

-- 
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System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-14 Thread Gareth Blades


On 11/10/13 18:43, Tiago Geada wrote:

Hi,

Seems a great workaround from Gareth Blades. Thanks I will try it.

Any way to make asterisk log a line in /var/log/messages ?

I normally have all the verbose output sent to the log file so anything 
in the NoOp() line gets logged to the file so thats what I use.
You could use the Log() or Verbose() applications if you only have 
errors written to the file as with those commands you can specify a log 
level.
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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-11 Thread Tiago Geada

Hi,

Seems a great workaround from Gareth Blades. Thanks I will try it.

Any way to make asterisk log a line in /var/log/messages ?


On 10 October 2013 19:44, Michelle Dupuis mdup...@ocg.ca wrote:

  Gareth:

 Did you check if your message (or security) log recorded anything during
 these attempts?  If so, can you post the content of the logs during this
 attack?

 M
  --
 *From:* asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad [
 asghar...@gmail.com]
 *Sent:* Tuesday, October 01, 2013 11:53 AM
 *To:* Asterisk Users List
 *Subject:* Re: [asterisk-users] Failed to authenticate user
 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

   Hi,
 Bad boys trying to guess a valid username.
 in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 1st
 invite.


 On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades 
 mailinglist+aster...@dns99.co.uk wrote:

 On 01/10/13 15:44, gincantalupo wrote:

 On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo 
 gincantal...@fgasoftware.com wrote:

 Hi,

 I get a lot of these messages on my Asterisk CLI:

 Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS
 ;tag=03f82bb9

 as if my PBX machine is trying to authenticate to itself. It seems
 someone is attacking my asterisk PBX.

 Is there a way to fix this problem?


 in sip.conf I have guest connections permitted and have them going to the
 default context which contains :-

 [default]
 ; all unauthenticated connection attempts from the internet come in here.
 exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
 ${SIP_HEADER(Contact)})
 exten = _[+*#0-9].,n,Congestion

 Then in fail2ban I have it match the following :-

 failregex = Registration from .* failed for \'HOST\' - Wrong password
 Unauthenticated call attempt .*\@HOST\:


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-10 Thread Michelle Dupuis

Gareth:

Did you check if your message (or security) log recorded anything during these 
attempts?  If so, can you post the content of the logs during this attack?

M

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Asghar Mohammad 
[asghar...@gmail.com]
Sent: Tuesday, October 01, 2013 11:53 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] Failed to authenticate user 
1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 1st 
invite.


On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades 
mailinglist+aster...@dns99.co.ukmailto:mailinglist+aster...@dns99.co.uk 
wrote:
On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo 
gincantal...@fgasoftware.commailto:gincantal...@fgasoftware.com wrote:
Hi,

I get a lot of these messages on my Asterisk CLI:

Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9

as if my PBX machine is trying to authenticate to itself. It seems someone is 
attacking my asterisk PBX.

Is there a way to fix this problem?

in sip.conf I have guest connections permitted and have them going to the 
default context which contains :-

[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt - 
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion

Then in fail2ban I have it match the following :-

failregex = Registration from .* failed for \'HOST\' - Wrong password
Unauthenticated call attempt .*\@HOST\:


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-02 Thread gincantalupo


Hi Garet,

ok but since the messages contain my own public IP with this method I'm 
banning my public IP not the real attacker IP. Am I wrong?


Giorgio


On 10/01/2013 05:26 PM, Gareth Blades wrote:

On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo 
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com 
wrote:


Hi,

I get a lot of these messages on my Asterisk CLI:

Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9

as if my PBX machine is trying to authenticate to itself. It
seems someone is attacking my asterisk PBX.

Is there a way to fix this problem?



in sip.conf I have guest connections permitted and have them going to 
the default context which contains :-


[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt - 
${SIP_HEADER(Contact)})

exten = _[+*#0-9].,n,Congestion

Then in fail2ban I have it match the following :-

failregex = Registration from .* failed for \'HOST\' - Wrong password
Unauthenticated call attempt .*\@HOST\:



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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-02 Thread gincantalupo


Hi Asghar,

surely this can improve security but what I'm looking for is something 
to find the real attacker IP address and ban it. Fail2ban bans my own 
public ip address.


Thank you

Giorgio


On 10/01/2013 05:53 PM, Asghar Mohammad wrote:

Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 
1st invite.



On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades 
mailinglist+aster...@dns99.co.uk 
mailto:mailinglist+aster...@dns99.co.uk wrote:


On 01/10/13 15:44, gincantalupo wrote:

On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.com
mailto:gincantal...@fgasoftware.com wrote:

Hi,

I get a lot of these messages on my Asterisk CLI:

Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9

as if my PBX machine is trying to authenticate to itself. It
seems someone is attacking my asterisk PBX.

Is there a way to fix this problem?



in sip.conf I have guest connections permitted and have them going
to the default context which contains :-

[default]
; all unauthenticated connection attempts from the internet come
in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
${SIP_HEADER(Contact)})
exten = _[+*#0-9].,n,Congestion

Then in fail2ban I have it match the following :-

failregex = Registration from .* failed for \'HOST\' - Wrong
password
Unauthenticated call attempt .*\@HOST\:


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-02 Thread Gareth Blades


On 02/10/13 16:13, gincantalupo wrote:

Hi Garet,

ok but since the messages contain my own public IP with this method 
I'm banning my public IP not the real attacker IP. Am I wrong?


Giorgio


No the asterisk dialplan entry is pulling the IP address out of the SIP 
Contact: header which in the attacks we have seen always seems to be the 
correct IP address.



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[asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-01 Thread gincantalupo


Hi,

I get a lot of these messages on my Asterisk CLI:

Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9

as if my PBX machine is trying to authenticate to itself. It seems 
someone is attacking my asterisk PBX.


Is there a way to fix this problem?

Thank you.

Giorgio Incantalupo


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-01 Thread Ricardo Saffi Marques

Well, you could use some software like denyhosts or fail2ban to block an IP
after a predefined number of (failed) authentication attempts.

Regards,

Ricardo


On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo
gincantal...@fgasoftware.comwrote:

 Hi,

 I get a lot of these messages on my Asterisk CLI:

 Failed to authenticate user 1000sip:1000@MY_OWN_IP_**
 ADDRESS;tag=03f82bb9

 as if my PBX machine is trying to authenticate to itself. It seems someone
 is attacking my asterisk PBX.

 Is there a way to fix this problem?

 Thank you.

 Giorgio Incantalupo


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-01 Thread gincantalupo


Hi Ricardo,

we are already using fail2ban but it bans my own ip address not the real 
original ip of the attacker. How can I find it?


Thank you

Giorgio


On 10/01/2013 02:16 PM, Ricardo Saffi Marques wrote:
Well, you could use some software like denyhosts or fail2ban to block 
an IP after a predefined number of (failed) authentication attempts.


Regards,

Ricardo


On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo 
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com 
wrote:


Hi,

I get a lot of these messages on my Asterisk CLI:

Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9

as if my PBX machine is trying to authenticate to itself. It seems
someone is attacking my asterisk PBX.

Is there a way to fix this problem?

Thank you.

Giorgio Incantalupo


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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-01 Thread Gareth Blades


On 01/10/13 15:44, gincantalupo wrote:
On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo 
gincantal...@fgasoftware.com mailto:gincantal...@fgasoftware.com 
wrote:


Hi,

I get a lot of these messages on my Asterisk CLI:

Failed to authenticate user
1000sip:1000@MY_OWN_IP_ADDRESS;tag=03f82bb9

as if my PBX machine is trying to authenticate to itself. It seems
someone is attacking my asterisk PBX.

Is there a way to fix this problem?



in sip.conf I have guest connections permitted and have them going to 
the default context which contains :-


[default]
; all unauthenticated connection attempts from the internet come in here.
exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt - 
${SIP_HEADER(Contact)})

exten = _[+*#0-9].,n,Congestion

Then in fail2ban I have it match the following :-

failregex = Registration from .* failed for \'HOST\' - Wrong password
Unauthenticated call attempt .*\@HOST\:

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Re: [asterisk-users] Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS; tag=03f82bb9

2013-10-01 Thread Asghar Mohammad

Hi,
Bad boys trying to guess a valid username.
in sip.conf uncomment  alwaysauthreject=yes and Asterisk always reject 1st
invite.


On Tue, Oct 1, 2013 at 5:26 PM, Gareth Blades 
mailinglist+aster...@dns99.co.uk wrote:

  On 01/10/13 15:44, gincantalupo wrote:

 On Tue, Oct 1, 2013 at 5:07 AM, gincantalupo gincantal...@fgasoftware.com
  wrote:

 Hi,

 I get a lot of these messages on my Asterisk CLI:

 Failed to authenticate user 1000sip:1000@MY_OWN_IP_ADDRESS
 ;tag=03f82bb9

 as if my PBX machine is trying to authenticate to itself. It seems
 someone is attacking my asterisk PBX.

 Is there a way to fix this problem?


 in sip.conf I have guest connections permitted and have them going to the
 default context which contains :-

 [default]
 ; all unauthenticated connection attempts from the internet come in here.
 exten = _[+*#0-9].,1,NoOp(Unauthenticated call attempt -
 ${SIP_HEADER(Contact)})
 exten = _[+*#0-9].,n,Congestion

 Then in fail2ban I have it match the following :-

 failregex = Registration from .* failed for \'HOST\' - Wrong password
 Unauthenticated call attempt .*\@HOST\:


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Re: [asterisk-users] Failed to authenticate device Ext 110

2013-05-22 Thread Matthew J. Roth

asterisk users wrote:
 
 Registration trace
 (note that extension 88 is the voicemail extension, which the phone registers
 to also for MWI)
 -- http://pastebin.com/c3H700wa

There are no REGISTER requests in that trace.  All I see are SUBSCRIBE, NOTIFY,
OPTIONS, and INVITE dialogs.

 Call trace: 
 |Time | 10.8.0.6 | 
 | | | 192.168.6.2 | 
 |268.693661| INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP 
 From: Ext 110  sip:110@192.168.6.2 To: sip:88@192.168.6.2 
 | |(1024) -- (5060) | 
 |268.694449| 401 Unauthorized |SIP Status 
 | |(1024) -- (5060) | 
 |268.914195| ACK | |SIP Request 
 | |(1024) -- (5060) | 
 |268.945115| INVITE SDP (g711U g729 g722 telephone-eventRTP...e-101) |SIP 
 From: Ext 110  sip:110@192.168.6.2 To: sip:88@192.168.6.2 
 | |(1024) -- (5060) | 
 |268.945717| 403 Forbidden |SIP Status 
 | |(1024) -- (5060) | 
 |269.041417| ACK | |SIP Request 
 | |(1024) -- (5060) | 

This is just a failed INVITE probably due to the username and/or password being
incorrect.  It's also possible that bad ACLs (see the 'permit/deny/acl' settings
in sip.conf) could be to blame.  It's hard to say without seeing a full SIP
trace and Asterisk CLI output.

 I'm also confused by the reference in sip show peers to port 5062, as I
 can't see that anywhere in the configuration of either the phone or in
 sip.conf. All the other phones show port 5060 in the sip show peers output. 

Start there and work through the obvious issues one by one.  First, figure out
why the phone is showing up on port 5062 and correct it if necessary.  Then,
double-check the username and password.  Keep going down that path until it
leads to a resolution or report back to the list if you run into a roadblock.

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Failed to authenticate device Ext 110

2013-05-21 Thread asterisk users

I'm having a strange problem recently with a Yealink SIP-T28P phone
connected to Asterisk 11.4.0 via openvpn.  It was working fine for months,
and now when I dial anything from the phone, it shows Forbidden, and the
Asterisk console shows:

[May 21 10:47:49] NOTICE[28518][C-0004]: chan_sip.c:25189
handle_request_invite: Failed to authenticate device Ext 110 
sip:110@192.168.6.2;tag=1130259112

Asterisk 192.168.6.2
OpenVPN on router 10.8.0.1
Remote Yealink phone 10.8.0.6

The remote phone shows as being registered:
PBX*CLI sip show peers
Name/username  Host  Dyn Forcerport ACL Port Status  Description
110/110   10.8.0.6  D   A  5062   OK (111 ms) Yealink OpenVPN

Also, if there is voicemail in the mailbox for 110, the phone's message
light is lit and it beeps periodically.

toshi*CLI sip show peer 110


  * Name   : 110
  Description  : Yealink OpenVPN
  Secret   : Set
  MD5Secret: Not set
  Remote Secret: Not set
  Context  : remote-phones
  Record On feature : automon
  Record Off feature : automon
  Subscr.Cont. : Not set
  Language :
  Tonezone : Not set
  AMA flags: Unknown
  Transfer mode: open
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup: 1
  Pickupgroup  : 1
  Named Callgr :
  Nam. Pickupgr:
  MOH Suggest  :
  Mailbox  : 110
  VM Extension : asterisk
  LastMsgsSent : 1/0
  Call limit   : 4
  Max forwards : 0
  Dynamic  : Yes
  Callerid : Ext 110 110
  MaxCallBR: 384 kbps
  Expire   : 608
  Insecure : no
  Force rport  : No
  Symmetric RTP: No
  ACL  : Yes
  DirectMedACL : No
  T.38 support : No
  T.38 EC mode : Unknown
  T.38 MaxDtgrm: -1
  DirectMedia  : No
  PromiscRedir : No
  User=Phone   : No
  Video Support: Yes
  Text Support : No
  Ign SDP ver  : No
  Trust RPID   : Yes
  Send RPID: Yes
  Subscriptions: Yes
  Overlap dial : No
  DTMFmode : rfc2833
  Timer T1 : 500
  Timer B  : 32000
  ToHost   :
  Addr-IP : 10.8.0.6:5062
  Defaddr-IP  : 10.8.0.6:5060
  Prim.Transp. : UDP
  Allowed.Trsp : UDP
  Def. Username: 110
  SIP Options  : (none)
  Codecs   : (ulaw)
  Codec Order  : (ulaw:20)
  Auto-Framing :  No
  Status   : OK (237 ms)
  Useragent: Yealink SIP-T28P 2.61.23.3 00:15:65:xx.xx.xx
  Reg. Contact : sip:110@10.8.0.6:5062
  Qualify Freq : 6 ms
  Keepalive: 0 ms
  Sess-Timers  : Accept
  Sess-Refresh : uas
  Sess-Expires : 1800 secs
  Min-Sess : 90 secs
  RTP Engine   : asterisk
  Parkinglot   :
  Use Reason   : No
  Encryption   : No

sip.conf:

[110]
context=remote-phones
type=peer
host=dynamic
qualify=1500
canreinvite=no
dtmfmode=rfc2833
progressinband=no
callgroup=1
pickupgroup=1   ; We can do call pickup for call group 1
call-limit=4
busy-level=1
qualify=yes
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0
nat=no
qualify=8000
description=Yealink OpenVPN
defaultuser=110
secret=x
callerid=Ext 110 110
mailbox=110
defaultip=10.8.0.6
port=5060
disallow=all
allow=ulaw

Any suggestions on what might be happening here, and how it could be
resolved?

THANKS ALL!
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Re: [asterisk-users] Failed to authenticate device Ext 110

2013-05-21 Thread Matthew J. Roth

asterisk users wrote:
 
 I'm having a strange problem recently with a Yealink SIP-T28P phone connected
 to Asterisk 11.4.0 via openvpn. It was working fine for months, and now when I
 dial anything from the phone, it shows Forbidden, and the Asterisk console
 shows: 
 
 [May 21 10:47:49] NOTICE[28518][C-0004]: chan_sip.c:25189 
 handle_request_invite: Failed to authenticate device Ext 110  
 sip:110@192.168.6.2 ;tag=1130259112 
 
 Asterisk 192.168.6.2 
 OpenVPN on router 10.8.0.1 
 Remote Yealink phone 10.8.0.6 
 
 The remote phone shows as being registered: 
 PBX*CLI sip show peers 
 Name/username Host Dyn Forcerport ACL Port Status Description 
 110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN 
 
 Also, if there is voicemail in the mailbox for 110, the phone's message light
 is lit and it beeps periodically. 
 
 ...
 
 Any suggestions on what might be happening here, and how it could be 
 resolved? 


That is quite strange.  Please provide SIP traces of the dialogs between
Asterisk and the phone in the following two scenarios:

  1) Phone registering to Asterisk (presumably successful)
  2) Phone dialing to Asterisk (presumably unsuccessful)

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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Re: [asterisk-users] Failed to authenticate device Ext 110

2013-05-21 Thread asterisk users

On Tue, May 21, 2013 at 11:26 AM, Matthew J. Roth mr...@imminc.com wrote:

 asterisk users wrote:
 
  I'm having a strange problem recently with a Yealink SIP-T28P phone
 connected
  to Asterisk 11.4.0 via openvpn. It was working fine for months, and now
 when I
  dial anything from the phone, it shows Forbidden, and the Asterisk
 console
  shows:
 
  [May 21 10:47:49] NOTICE[28518][C-0004]: chan_sip.c:25189
 handle_request_invite: Failed to authenticate device Ext 110 
 sip:110@192.168.6.2 ;tag=1130259112
 
  Asterisk 192.168.6.2
  OpenVPN on router 10.8.0.1
  Remote Yealink phone 10.8.0.6
 
  The remote phone shows as being registered:
  PBX*CLI sip show peers
  Name/username Host Dyn Forcerport ACL Port Status Description
  110/110 10.8.0.6 D A 5062 OK (111 ms) Yealink OpenVPN
 
  Also, if there is voicemail in the mailbox for 110, the phone's message
 light
  is lit and it beeps periodically.
 
  ...
 
  Any suggestions on what might be happening here, and how it could be
 resolved?


 That is quite strange.  Please provide SIP traces of the dialogs between
 Asterisk and the phone in the following two scenarios:

   1) Phone registering to Asterisk (presumably successful)
   2) Phone dialing to Asterisk (presumably unsuccessful)

 Regards,

 Matthew Roth
 InterMedia Marketing Solutions
 Software Engineer and Systems Developer

 --


Registration trace
(note that extension 88 is the voicemail extension, which the phone
registers to also for MWI)
-- http://pastebin.com/c3H700wa

Call trace:
|Time | 10.8.0.6  |
| |   | 192.168.6.2   |
|268.693661| INVITE SDP (g711U g729 g722
telephone-eventRTP...e-101)  |SIP From: Ext 110 
sip:110@192.168.6.2 To:sip:88@192.168.6.2
| |(1024)   --  (5060)   |
|268.694449| 401 Unauthorized  |SIP Status
| |(1024)   --  (5060)   |
|268.914195| ACK   |   |SIP Request
| |(1024)   --  (5060)   |
|268.945115| INVITE SDP (g711U g729 g722
telephone-eventRTP...e-101)  |SIP From: Ext 110 
sip:110@192.168.6.2 To:sip:88@192.168.6.2
| |(1024)   --  (5060)   |
|268.945717| 403 Forbidden |SIP Status
| |(1024)   --  (5060)   |
|269.041417| ACK   |   |SIP Request
| |(1024)   --  (5060)   |


I'm also confused by the reference in sip show peers to port 5062, as I
can't see that anywhere in the configuration of either the phone or in
sip.conf.  All the other phones show port 5060 in the sip show peers
output.
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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

2012-01-04 Thread Jayesh Labade

Please help me..

Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08]
 NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i registered
 this trunk in softphone like Xlite, there is no problem with outbound
 calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com


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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

2012-01-04 Thread virendra bhati

Hi,

Give the complete details about the asterisk version, and SIP trunk conf
details


On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08]
 NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i registered
 this trunk in softphone like Xlite, there is no problem with outbound
 calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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Software Engineer
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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

2012-01-04 Thread Jayesh Labade

Hi,

I am using asterisk ver 1.8.8.1.

My SIP trunk conf details are below..

[general]
context=default ; Default context for incoming calls
realm=192.168.1.55
allowguest=yes
realmauth=yes
send_rpid=pai

register = test02:test02@192.168.1.55


[test02]
type=peer
nat=no
canreinvite=no
host=192.168.1.55
;realm=test02@192.168.1.55
context=incoming
secret=test02
permit=192.168.1.0/255.255.255.0
username=test02
fromuser=test02
fromdomain=192.168.1.55
defaultuser=test02
insecure=invite,port
outboundproxy=192.168.1.55
promiscredir=yes
userphone=yes

For more details you can find my paste in pastebin.. Links given below.

While Dialing call fro Xlite send following Sip header F=
sip:test02@192.168.1.55. And if tried to register same account in asterisk
trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why
asterisk sends anonymous.invalid instead of domain name..Help me

Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote:

 Hi,

 Give the complete details about the asterisk version, and SIP trunk conf
 details


 On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 
 jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3
 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i registered
 this trunk in softphone like Xlite, there is no problem with outbound
 calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

2012-01-04 Thread virendra bhati

Hi checked your debug like.

Did you check that your SIP device ir registered with server ?
if yes then dial below command from CLI

*originate sip/test02 application dial*



On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hi,

 I am using asterisk ver 1.8.8.1.

 My SIP trunk conf details are below..

 [general]
 context=default ; Default context for incoming calls
 realm=192.168.1.55
 allowguest=yes
 realmauth=yes
 send_rpid=pai

 register = test02:test02@192.168.1.55


 [test02]
 type=peer
 nat=no
 canreinvite=no
 host=192.168.1.55
 ;realm=test02@192.168.1.55
 context=incoming
 secret=test02
 permit=192.168.1.0/255.255.255.0
 username=test02
 fromuser=test02
 fromdomain=192.168.1.55
 defaultuser=test02
 insecure=invite,port
 outboundproxy=192.168.1.55
 promiscredir=yes
 userphone=yes

 For more details you can find my paste in pastebin.. Links given below.

 While Dialing call fro Xlite send following Sip header F=
 sip:test02@192.168.1.55. And if tried to register same account in
 asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont
 know why asterisk sends anonymous.invalid instead of domain name..Help me


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote:

 Hi,

 Give the complete details about the asterisk version, and SIP trunk conf
 details


 On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 
 jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3
 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i
 registered this trunk in softphone like Xlite, there is no problem with
 outbound calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

2012-01-04 Thread Jayesh Labade

Hi virendra,

Dialed same command.. I got below output

ast18*CLI originate sip/test02 application dial
  == Using SIP RTP CoS mark 5
[Jan  4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite:
Failed to authenticate on INVITE to 'Anonymous
sip:test02@anonymous.invalid:192;tag=as417a5527'


Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati virbh...@gmail.com wrote:

 Hi checked your debug like.

 Did you check that your SIP device ir registered with server ?
 if yes then dial below command from CLI

 *originate sip/test02 application dial*




 On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hi,

 I am using asterisk ver 1.8.8.1.

 My SIP trunk conf details are below..

 [general]
 context=default ; Default context for incoming calls
 realm=192.168.1.55
 allowguest=yes
 realmauth=yes
 send_rpid=pai

 register = test02:test02@192.168.1.55


 [test02]
 type=peer
 nat=no
 canreinvite=no
 host=192.168.1.55
 ;realm=test02@192.168.1.55
 context=incoming
 secret=test02
 permit=192.168.1.0/255.255.255.0
 username=test02
 fromuser=test02
 fromdomain=192.168.1.55
 defaultuser=test02
 insecure=invite,port
 outboundproxy=192.168.1.55
 promiscredir=yes
 userphone=yes

 For more details you can find my paste in pastebin.. Links given below.

 While Dialing call fro Xlite send following Sip header F=
 sip:test02@192.168.1.55. And if tried to register same account in
 asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I
 dont know why asterisk sends anonymous.invalid instead of domain name..Help
 me


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.comwrote:

 Hi,

 Give the complete details about the asterisk version, and SIP trunk conf
 details


 On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade 
 jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
  wrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3
 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed 
 to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i
 registered this trunk in softphone like Xlite, there is no problem with
 outbound calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

2012-01-04 Thread sean darcy


On 1/4/2012 4:37 AM, Jayesh Labade wrote:

Please help me..

Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
mailto:jayesh.lab...@gmail.com wrote:

Hello Experts,

I have pasted my issue in http://pastebin.com/zBGVmdcY

I Cant able to Originate call from SIp trunk..I got this [Jan 3
11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite:
Failed to authenticate on INVITE to 'Anonymous
sip:test02@anonymous.invalid;tag=as57d3a806'
i am unable to make outbound call from this trunk. while if i
registered this trunk in softphone like Xlite, there is no problem
with outbound calls. Help me.

please find sip.conf file in http://pastebin.com/zBGVmdcY

I have pasted sip debug with verbosity of failed call
http://pastebin.com/jL2ki0s8


Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com




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Try:
register = test02:test02@192.168.1.55/s

sean



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[asterisk-users] - Failed to authenticate user

2008-05-19 Thread aby azid

Hie,

I managed to connect two Asterisk box via SIP. My problem is when I login
using Realtime SIP, I will get

 chan_sip.c:8373 check_auth: username mismatch, have 8003000777, digest
has voip3

Failed to authenticate user 8003000777
sip:[EMAIL PROTECTED][EMAIL PROTECTED]
;tag=as4d11916d

when trying to send call to the 2nd server.

but when I login using sip.conf setting I managed to send call to the 2nd
server with no errors. Any ideas?

cheers
Aby
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Re: [asterisk-users] Failed to authenticate on INVITE

2007-04-20 Thread Rilawich Ango


I use realtime.  Both information and extensions are stored in DB.  It
is just a simple setting of the user with dial plan Dial([EMAIL PROTECTED]).
exten = 9003,1,Dial([EMAIL PROTECTED])
What I found is the following.

9002 --- S1 --- S2
9002 can make request to S1 and S1 forward the request to S2.
9002 --- S1 --- S2
S2 returns the mentioned error message to S1.  (What I guess is 9002
only registers in S1 not in S2, so mentioned error message issued by
S2).

It is what I got from the above case.  Do you have such configuration?
I have no idea to solve the problem

On 4/20/07, dave cantera [EMAIL PROTECTED] wrote:

ango,
can you provide some sip.conf and extens.conf info?
daveC

Rilawich Ango wrote:
 hi,
  I have 2 asterisks with the following configuration.
 asterisk server 1 (S1) has an user 9002
 asterisk server 2 (S2) has an user 9003
 Both users can make call to each other without problem.
 Now I add both users to both servers, i.e.
 asterisk server 1 (S1) has users 9002,9003
 asterisk server 2 (S2) has users 9002,9003
 When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa.  Both 
processes
 failed to make call with the following error.
 Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite:
 Failed to authenticate on INVITE to '9002
 sip:[EMAIL PROTECTED];tag=as2ff0c493'
 Any solution to let them call each others?
 ango
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Re: [asterisk-users] Failed to authenticate on INVITE

2007-04-20 Thread dave cantera





ango, 
I have been playing with connecting two * servers... I had to stop but
I do think I had it working... even with this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
it wasn't as straight forward as I would have liked... I used a
register on one box and a conf entry on the other. then I reversed the
config for the other * box

pbx82 = 10.10.15.82
pbx15 = 10.10.15.15

on pbx15

sip.conf
register = sip_pbx15:[EMAIL PROTECTED]
[sip_to_pbx82]
type=user
username=sip_pbx15
accountcode=sip_from_pbx15
secret=1234
context=sip_from_pbx15
host=10.10.15.82
disallow=all
allow=ulaw
allow=alaw
allow=gsm

extensions.conf
[sip_pbx15_to_pbx82]
; dial a pbx82 extension via SIP with 982XXX where XXX is the extension
exten =
_982XXX,1,Dial(SIP/sip_pbx15:[EMAIL PROTECTED]/${EXTEN:3},20,r)
;exten = _982XXX,1,Dial(SIP/${EXTEN:3},20,r)
exten = _982XXX,n,Playback(connection-failed)
exten = _982XXX,n,Playback(vm-goodbye)
exten = _982XXX,n,Congestion
exten = _982XXX,n,Hangup

on pbx82

extensions.conf
[sip_from_pbx15]
exten = _XXX,1,Wait(1)
exten = _XXX,n,Answer()
exten = _XXX,n,Dial(SIP/${EXTEN},20,,r)
exten = _XXX,n,VoiceMailMain
exten = _XXX,n,Hangup()

[sip_from_pbx15] must be accessible in your inbound or default
context...
I don't think I made any general section changes...

it has been a few weeks since I played with it and I went only one
way... but if it worked one way it should work the other way too by
reverse duplicating the above config on pbx82 and pbx15 respectively.
let me know how you make out...
daveC


Rilawich Ango wrote:
I use realtime. Both information and extensions are
stored in DB. It
  
is just a simple setting of the user with dial plan "Dial([EMAIL PROTECTED])".
  
exten = 9003,1,Dial([EMAIL PROTECTED])
  
What I found is the following.
  
  
9002 --- S1 --- S2
  
9002 can make request to S1 and S1 forward the request to S2.
  
9002 --- S1 --- S2
  
S2 returns the mentioned error message to S1. (What I guess is 9002
  
only registers in S1 not in S2, so mentioned error message issued by
  
S2).
  
  
It is what I got from the above case. Do you have such configuration?
  
I have no idea to solve the problem
  
  
On 4/20/07, dave cantera [EMAIL PROTECTED] wrote:
  
  ango,

can you provide some sip.conf and extens.conf info?

daveC


Rilawich Ango wrote:

 hi,

 I have 2 asterisks with the following configuration.

 asterisk server 1 (S1) has an user 9002

 asterisk server 2 (S2) has an user 9003

 Both users can make call to each other without problem.

 Now I add both users to both servers, i.e.

 asterisk server 1 (S1) has users 9002,9003

 asterisk server 2 (S2) has users 9002,9003

 When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa. Both
processes

 failed to make call with the following error.

 Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802
handle_response_invite:

 Failed to authenticate on INVITE to '"9002"

 sip:[EMAIL PROTECTED];tag=as2ff0c493'

 Any solution to let them call each others?

 ango

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[asterisk-users] Failed to authenticate on INVITE

2007-04-19 Thread Rilawich Ango


hi,
 I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2 (S2) has users 9002,9003
When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa.  Both processes
failed to make call with the following error.
Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite:
Failed to authenticate on INVITE to '9002
sip:[EMAIL PROTECTED];tag=as2ff0c493'
Any solution to let them call each others?
ango
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Re: [asterisk-users] Failed to authenticate on INVITE

2007-04-19 Thread dave cantera


ango,
can you provide some sip.conf and extens.conf info?
daveC

Rilawich Ango wrote:

hi,
 I have 2 asterisks with the following configuration.
asterisk server 1 (S1) has an user 9002
asterisk server 2 (S2) has an user 9003
Both users can make call to each other without problem.
Now I add both users to both servers, i.e.
asterisk server 1 (S1) has users 9002,9003
asterisk server 2 (S2) has users 9002,9003
When 9002 dials 9003, Dial(SIP/[EMAIL PROTECTED]) or visa versa.  Both processes
failed to make call with the following error.
Apr 16 11:55:41 NOTICE[19658]: chan_sip.c:9802 handle_response_invite:
Failed to authenticate on INVITE to '9002
sip:[EMAIL PROTECTED];tag=as2ff0c493'
Any solution to let them call each others?
ango
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[asterisk-users] failed to authenticate on invite

2006-11-07 Thread Damon Estep









I have 2 asterisk boxes connected via SIP



box 1 sip peer connected
to box 2 (ip addresses intentionally removed)



[ast20]

type=friend

host=x.x.x.20

insecure=very

context=subscriber

dtmfmode=inband

qualify=no

canreinvite=no

disallow=all

allow=ulaw



box 2 sip peer connected
to box 1



[sbb19]

type=friend

host=64.1.8.19

insecure=very

context=inbound

dtmfmode=inband

qualify=yes

canreinvite=no

disallow=all

allow=ulaw



I then have 2 UAs registed on box 1, both have identical
configs with the exception of username, but one is a Polycom IP501 and the
other is a Linksys PAP2



The IP 501 can call to box 2 with no issues, also calls
originated on a PRI connected to box 1 connect to box 2 with no issues.



The Linksys UA can not call box 2, here is the error
(numbers intentionally removed);



-- Executing dial(SIP/##0850-b6669f58,
SIP/[EMAIL PROTECTED])

 -- Called [EMAIL PROTECTED]

Nov 7 07:20:45 NOTICE[21059]:
chan_sip.c:9709 handle_response_invite: Failed to authenticate on INVITE to
'name removed sip:[EMAIL PROTECTED];tag=as38826922'

 -- SIP/ast20-09c8b110
is circuit-busy

 == Everyone is busy/congested
at this time (1:0/1/0)



I have looked at sip debugs from
both scenarios, and the invites from box 1 to box 2 look nearly identical, box
2 never shows the call when it fails.



I am assuming that there is
something that needs to be changed on the ATA or peer config to get it to be
able to call via box1 to box2 without requiring authentication, but can not
figure out what.



Any ideas?












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[Asterisk-Users] Failed to authenticate user

2005-08-08 Thread Neil Bullock

Please can anyone help? I'm trying to setup our asterisk system to allow
any user to place a call regardless of whether they are registered or
not. Ideally I want to place unregistered callers into a specific
context but authentication is based then on the number they are dialing
rather than having them registered to the server.

Hope this makes some sense!

Cheers,

Neil

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[Asterisk-Users] Failed to authenticate

2005-04-24 Thread lie ka


HI,all!
 I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these:
sip.conf
[general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes
dtmfmode=rfc2833canreinvite=no
context=defaulttos=0x18dbname=asteriskdbhost=localhostdbuser=asteriskdbpass=password

extensions.conf
[general]static=yeswriteprotect=no
[globals]CONSOLE=Console/dsp

[local]
exten = _X.,1,Dial(SIP/${EXTEN},20,t)exten = _X.,2,Hangup

[default]include = demoinclude = local

I have also setted callidnum 1000-1010 in mysql database.First,it can dial out and receive a call well.(in internal) then I alter callidnum 1000 to 
1000.It can registered successfully and it can receive a call ,but it cannot dial out .There are some words in my asterisk console:"Failed to authenticate user "aaa" sip:[EMAIL PROTECTED]; tag=164262242".So,I tried change callidnum to 1000, it works. I don't know what happen.Can anybody tell me what's the matter ? thanks!

in addition,If I don't use sipfriends with mysql, it does well !

Do You Yahoo!?
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[Asterisk-Users] Failed to authenticate

2005-04-23 Thread lie ka


HI,all!
 I have setted up an Asterisk server on my linux.My sip.conf and extensions.conf configurations like these:
sip.conf
[general]port=5060disallow=allallow=alawallow=ulawallow=gsmnat=yes
dtmfmode=rfc2833canreinvite=no
context=defaulttos=0x18dbname=asteriskdbhost=localhostdbuser=asteriskdbpass=password

extensions.conf
[general]static=yeswriteprotect=no
[globals]CONSOLE=Console/dsp

[local]
exten = _X.,1,Dial(SIP/${EXTEN},20,t)exten = _X.,2,Hangup

[default]include = demoinclude = local

I have also setted callidnum 1000-1010 in mysql database.First,it can dial out and receive a call well.(in internal) then I alter callidnum 1000 to 
1000.It can registered successfully and it can receive a call ,but it cannot dial out .There are some words in my asterisk console:"Failed to authenticate user "aaa" sip:[EMAIL PROTECTED]; tag=164262242".So,I tried change callidnum to 1000, it works. I don't know what happen.Can anybody tell me what's the matter ? thanks!

Do You Yahoo!?
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[Asterisk-Users] Failed to authenticate on INVITE to '601 ...

2004-10-24 Thread Ronald Wiplinger

I have installed the first time Asterisk,  (forgive me simple questions)
I have also installed the demo.
After testing demo (call 1000, call 600, ...) I changed in the 
extensions.conf:

; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
exten = 5552220,1,Dial(SIP/601,20,r)
exten = 4782156,1,Dial(SIP/602,20,r)
[sipgate.de]
exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten = _0049X.,2,Playback(invalid)
exten = _0049X.,3,Hangup
[sipgate.co.uk]
exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _0044X.,2,Playback(invalid)
exten = _0044X.,3,Hangup

in sip.conf I have:
register = 5552220:[EMAIL PROTECTED]/5552220
register = 4782156:[EMAIL PROTECTED]/4782156
[601]
type=friend
username=601
secret=pwd-601
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=601
nat=yes
caller-id=Ronald 1 601
[602]
type=friend
username=602
secret=pwd-602
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=602
nat=yes
caller-id=Ronald 2 602
[sipgate.de]
type=friend
username=5552220
secret=pwd-de
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=pwd-uk
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.co.uk
nat=yes
context=incomingsipgate
canreinvite=no


The console shows when I want to dial at sipgate.de  the number 1 
(test) or 5 (Voicemail):  00491

-- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new stack
-- Called [EMAIL PROTECTED]
Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to 
authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254'
-- Nobody picked up in 3 ms
-- Executing Playback(SIP/601-ea8b, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, 00491, 3) exited non-zero on 'SIP/601-ea8b'
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b'


What do I miss?
bye
Ronald
begin:vcard
fn:Ronald Wiplinger
n:Wiplinger;Ronald
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
version:2.1
end:vcard

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Re: [Asterisk-Users] Failed to authenticate on INVITE to '601 ... solved

2004-10-24 Thread Ronald Wiplinger

Ronald Wiplinger wrote:
I have installed the first time Asterisk,  (forgive me simple 
questions)

I have also installed the demo.

I solved it with the newest cvs version !!!
bye
Ronald

After testing demo (call 1000, call 600, ...) I changed in the 
extensions.conf:

; include = demo
include = incomingsipgate
include = sipgate.de
include = sipgate.col.uk
[incomingsipgate]
exten = 5552220,1,Dial(SIP/601,20,r)
exten = 4782156,1,Dial(SIP/602,20,r)
[sipgate.de]
exten = _0049X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten = _0049X.,2,Playback(invalid)
exten = _0049X.,3,Hangup
[sipgate.co.uk]
exten = _0044X.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
exten = _0044X.,2,Playback(invalid)
exten = _0044X.,3,Hangup

in sip.conf I have:
register = 5552220:[EMAIL PROTECTED]/5552220
register = 4782156:[EMAIL PROTECTED]/4782156
[601]
type=friend
username=601
secret=pwd-601
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=601
nat=yes
caller-id=Ronald 1 601
[602]
type=friend
username=602
secret=pwd-602
canreinvite=no
host=dynamic
defaultip=61.220.121.19
dtmfmode=rfc2833
mailbox=602
nat=yes
caller-id=Ronald 2 602
[sipgate.de]
type=friend
username=5552220
secret=pwd-de
host=sipgate.de
fromuser=5552220
fromdomain=sipgate.de
nat=yes
context=incomingsipgate
canreinvite=no
[sipgate.co.uk]
type=friend
username=4782156
secret=pwd-uk
host=sipgate.co.uk
fromuser=4782156
fromdomain=sipgate.co.uk
nat=yes
context=incomingsipgate
canreinvite=no


The console shows when I want to dial at sipgate.de  the number 1 
(test) or 5 (Voicemail):  00491

-- Executing Dial(SIP/601-ea8b, SIP/[EMAIL PROTECTED]|30|r) in new 
stack
-- Called [EMAIL PROTECTED]
Oct 24 23:00:48 NOTICE[213005]: c6779 handle_response: Failed to 
authenticate on INVITE to '601 sip:[EMAIL PROTECTED];tag=as25c3e254'
-- Nobody picked up in 3 ms
-- Executing Playback(SIP/601-ea8b, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Got SIP response 481 Call Leg Does Not Exist back from 217.10.79.9
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, 00491, 3) exited non-zero on 
'SIP/601-ea8b'
-- Executing Hangup(SIP/601-ea8b, ) in new stack
== Spawn extension (default, h , 1) exited non-zero on 'SIP/601-ea8b'


What do I miss?
bye
Ronald
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(from USA dial (408)253-3153 # 7303)
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RE: [Asterisk-Users] Failed to authenticate on INVITE

2004-09-21 Thread Whisker, Peter




For 
info

The 
new chan_sip2.c and recent CVS (yesterday) fix this and I can now use Asterisk 
to make calls on the sip.btcommunicator.bt.net service. If anyone wants help 
withthe settings, e-mail me off list.

:)

Peter

-Original Message-From: Whisker, Peter 
[mailto:[EMAIL PROTECTED]Sent: 17 September 2004 
14:40To: 'Asterisk Users Mailing List - Non-Commercial 
Discussion'Subject: RE: [Asterisk-Users] Failed to authenticate on 
INVITE
I am 
getting this also.

I am 
trying to get Asterisk to talk similarly to BT Communicator to the BT server. I 
can register but then the INVITE fails.

BT are 
mixed up with theirdomains (in fact in the INVITE their software has a To: 
header withnumber@domain1 and an auth URI referencing 
number@domain2. The realm is domain1.) This can't be done in Asterisk 
where it is consistent about the URI.

I had 
been blaming this, but if you are having problems too...

I get 
the standard 407 header requesting Proxy Auth for the call. Asterisk submits the 
INVITE with auth and after the usual "Trying" I just get another 407. I have 
traces of Asterisk and the client which works and they seem so similar in what 
they do. I have made all the port ranges the same too. BT Communicator fails if 
you use port 5060 for the SIP client- they use 5052.

Peter



-Original Message-From: Stig Thune 
[mailto:[EMAIL PROTECTED]Sent: 17 September 2004 
12:55To: [EMAIL PROTECTED]Subject: 
[Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response: 
Failed to authenticate on INVITE to 
'sip:[EMAIL PROTECTED];tag=as0f1d3429'



sip.conf


register = 
1234:[EMAIL PROTECTED]





extension.conf
--

;; Own extensions;exten = 
0852509516,1,Goto(resepsjon-own,s,1)

;[resepsjon-own];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/choose) 
; Meny, 1 for support, 2 for support, 3 for wx3exten = 
s,6,Wait(1)exten = 
s,7,Background(own/choosenumber) 
; dialer pushes a # ,and being sent 
to.. 
; ip-phone must be picked up in ,2ms,tr or hangupexten 
= 1,1,Goto(privatanslutningar,s,1)exten = 
2,1,Goto(foretagsanslutningar,s,1)

; #=hangupexten = 
#,1,Playback(custom/no-key-registered)exten = 
#,2,Hangup

exten = 
t,1,Goto(#,1) ; If they take too 
long, give upexten = i,1,Playback(invalid) ; "That's not valid, 
try again" inmenu]

;[privatanslutningar];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/privatanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3 
; dialer pushes a # ,and being sent to..

exten = 1,1,Answerexten = 
1,2,Queue(help-privatanslutningar-queue)exten = 
2,1,Answerexten = 
2,2,Queue(order-privatanslutningar-queue)exten = 
3,1,Answerexten = 
3,2,Queue(info-privatanslutningar-queue)

; #=hangup;exten = 
#,1,Playback(custom/no-key-registered);exten = 
#,2,Hangup

exten = 
t,1,Queue(general-privatanslutningar-queue) 
; If they take too long, give upexten = 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]

;[foretagsanslutningar];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/foretagsanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3 
; dialer pushes a # ,and being sent 
to.. 
; ip-phone must be picked up in ,2ms,tr or hangupexten 
= 1,1,Answerexten = 
1,2,Queue(info-bedriftsanslutningar-queue)exten = 
2,1,Answerexten = 
2,2,Queue(help-bedriftsanslutningar-queue)exten = 
3,1,Answerexten = 
3,2,Queue(error-bedriftsanslutningar-queue)

; #=hangup;exten = 
#,1,Playback(custom/no-key-registered);exten = #,2,Hangup

exten = 
t,1,Queue(general-bedriftsanslutningar-queue) 
; If they take too long, give upexten = 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]
--


The call gets into queue, then... the other phone 
rings.. and when I pick up - I get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate 
on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429'

I know that the register = works.. I have checked with my SIP-provider, 
and they say that it is logged in.

What else can be wrong ?

/ Stig HenningThis e-mail and any attachment is for 
authorised use by the intended recipient(s) only. It may contain proprietary 
material, confidential information and/or be subject to legal privilege. It 
should not be copied, disclosed to, retained or used by, any other party. If you 
are not an intended recipient then please promptly delete this e-mail and any 
attachment and all copies and inform the sender. Thank you.

This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privi

[Asterisk-Users] Failed to authenticate on INVITE

2004-09-17 Thread Stig Thune




NOTICE[98310]: chan_sip.c:6638 handle_response: 
Failed to authenticate on INVITE to 
'sip:[EMAIL PROTECTED];tag=as0f1d3429'



sip.conf


register = 
1234:[EMAIL PROTECTED]





extension.conf
--

;; Own extensions;exten = 
0852509516,1,Goto(resepsjon-own,s,1)

;[resepsjon-own];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/choose) 
; Meny, 1 for support, 2 for support, 3 for wx3exten = 
s,6,Wait(1)exten = 
s,7,Background(own/choosenumber) 
; dialer pushes a # ,and being sent 
to.. 
; ip-phone must be picked up in ,2ms,tr or hangupexten 
= 1,1,Goto(privatanslutningar,s,1)exten = 
2,1,Goto(foretagsanslutningar,s,1)

; #=hangupexten = 
#,1,Playback(custom/no-key-registered)exten = 
#,2,Hangup

exten = 
t,1,Goto(#,1) ; If they take too 
long, give upexten = i,1,Playback(invalid) ; "That's not valid, 
try again" inmenu]

;[privatanslutningar];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/privatanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3 
; dialer pushes a # ,and being sent to..

exten = 1,1,Answerexten = 
1,2,Queue(help-privatanslutningar-queue)exten = 
2,1,Answerexten = 
2,2,Queue(order-privatanslutningar-queue)exten = 
3,1,Answerexten = 
3,2,Queue(info-privatanslutningar-queue)

; #=hangup;exten = 
#,1,Playback(custom/no-key-registered);exten = 
#,2,Hangup

exten = 
t,1,Queue(general-privatanslutningar-queue) 
; If they take too long, give upexten = 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]

;[foretagsanslutningar];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/foretagsanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3 
; dialer pushes a # ,and being sent 
to.. 
; ip-phone must be picked up in ,2ms,tr or hangupexten 
= 1,1,Answerexten = 
1,2,Queue(info-bedriftsanslutningar-queue)exten = 
2,1,Answerexten = 
2,2,Queue(help-bedriftsanslutningar-queue)exten = 
3,1,Answerexten = 
3,2,Queue(error-bedriftsanslutningar-queue)

; #=hangup;exten = 
#,1,Playback(custom/no-key-registered);exten = #,2,Hangup

exten = 
t,1,Queue(general-bedriftsanslutningar-queue) 
; If they take too long, give upexten = 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]
--


The call gets into queue, then... the other phone 
rings.. and when I pick up - I get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate 
on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429'

I know that the register = works.. I have checked with my SIP-provider, 
and they say that it is logged in.

What else can be wrong ?

/ Stig Henning
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RE: [Asterisk-Users] Failed to authenticate on INVITE

2004-09-17 Thread Whisker, Peter




I am 
getting this also.

I am 
trying to get Asterisk to talk similarly to BT Communicator to the BT server. I 
can register but then the INVITE fails.

BT are 
mixed up with theirdomains (in fact in the INVITE their software has a To: 
header withnumber@domain1 and an auth URI referencing 
number@domain2. The realm is domain1.) This can't be done in Asterisk 
where it is consistent about the URI.

I had 
been blaming this, but if you are having problems too...

I get 
the standard 407 header requesting Proxy Auth for the call. Asterisk submits the 
INVITE with auth and after the usual "Trying" I just get another 407. I have 
traces of Asterisk and the client which works and they seem so similar in what 
they do. I have made all the port ranges the same too. BT Communicator fails if 
you use port 5060 for the SIP client- they use 5052.

Peter



-Original Message-From: Stig Thune 
[mailto:[EMAIL PROTECTED]Sent: 17 September 2004 
12:55To: [EMAIL PROTECTED]Subject: 
[Asterisk-Users] Failed to authenticate on INVITE
NOTICE[98310]: chan_sip.c:6638 handle_response: 
Failed to authenticate on INVITE to 
'sip:[EMAIL PROTECTED];tag=as0f1d3429'



sip.conf


register = 
1234:[EMAIL PROTECTED]





extension.conf
--

;; Own extensions;exten = 
0852509516,1,Goto(resepsjon-own,s,1)

;[resepsjon-own];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/choose) 
; Meny, 1 for support, 2 for support, 3 for wx3exten = 
s,6,Wait(1)exten = 
s,7,Background(own/choosenumber) 
; dialer pushes a # ,and being sent 
to.. 
; ip-phone must be picked up in ,2ms,tr or hangupexten 
= 1,1,Goto(privatanslutningar,s,1)exten = 
2,1,Goto(foretagsanslutningar,s,1)

; #=hangupexten = 
#,1,Playback(custom/no-key-registered)exten = 
#,2,Hangup

exten = 
t,1,Goto(#,1) ; If they take too 
long, give upexten = i,1,Playback(invalid) ; "That's not valid, 
try again" inmenu]

;[privatanslutningar];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/privatanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3 
; dialer pushes a # ,and being sent to..

exten = 1,1,Answerexten = 
1,2,Queue(help-privatanslutningar-queue)exten = 
2,1,Answerexten = 
2,2,Queue(order-privatanslutningar-queue)exten = 
3,1,Answerexten = 
3,2,Queue(info-privatanslutningar-queue)

; #=hangup;exten = 
#,1,Playback(custom/no-key-registered);exten = 
#,2,Hangup

exten = 
t,1,Queue(general-privatanslutningar-queue) 
; If they take too long, give upexten = 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]

;[foretagsanslutningar];exten = 
s,1,Answerexten = s,2,SetMusicOnHold(default)exten = 
s,3,DigitTimeout,5exten = s,4,ResponseTimeout,15exten 
= 
s,5,Background(own/foretagsanslutningar) 
; Meny, 1 for support, 2 for support, 3 for 
wx3 
; dialer pushes a # ,and being sent 
to.. 
; ip-phone must be picked up in ,2ms,tr or hangupexten 
= 1,1,Answerexten = 
1,2,Queue(info-bedriftsanslutningar-queue)exten = 
2,1,Answerexten = 
2,2,Queue(help-bedriftsanslutningar-queue)exten = 
3,1,Answerexten = 
3,2,Queue(error-bedriftsanslutningar-queue)

; #=hangup;exten = 
#,1,Playback(custom/no-key-registered);exten = #,2,Hangup

exten = 
t,1,Queue(general-bedriftsanslutningar-queue) 
; If they take too long, give upexten = 
i,1,Playback(invalid) 
; "That's not valid, try again" inmenu]
--


The call gets into queue, then... the other phone 
rings.. and when I pick up - I get this message:
NOTICE[98310]: chan_sip.c:6638 handle_response: Failed to authenticate 
on INVITE to 'sip:[EMAIL PROTECTED];tag=as0f1d3429'

I know that the register = works.. I have checked with my SIP-provider, 
and they say that it is logged in.

What else can be wrong ?

/ Stig Henning

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Re: [Asterisk-Users] Failed to authenticate on INVITE

2004-06-17 Thread Jason Williams

At 16:49 16/06/2004 -0400, Eric wrote:
I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
These two boxes talk to eachother via sip, not iax.  Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box.  Removing the secret from each box's sip config seems to work but
is utterly braindead.
include the line in sip.conf for each user the call
insecure=yes   ; To match a peer based by IP address only 
and not peer

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Re: [Asterisk-Users] Failed to authenticate on INVITE

2004-06-17 Thread Eric Einhorn

Hi Jason,

Thanks for your reply.  I didn't really want to use the insecure option,
that defeats the purpose of using a password :)

I was, however, able to specify user= in my sip.conf entity and that
solved the problem I was having.

Thanks again.

- Eric



On Thu, 17 Jun 2004 10:17:54 +0100
Jason Williams [EMAIL PROTECTED] wrote:

 At 16:49 16/06/2004 -0400, Eric wrote:
 I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).
 
 These two boxes talk to eachother via sip, not iax.  Since the upgrade, I
 get the error Failed to authenticate on INVITE trying to make calls to/from
 either box.  Removing the secret from each box's sip config seems to work but
 is utterly braindead.
 
 include the line in sip.conf for each user the call
 
 insecure=yes   ; To match a peer based by IP address only 
 and not peer
 
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[Asterisk-Users] Failed to authenticate on INVITE

2004-06-16 Thread Eric Einhorn

Hi,

I upgraded my two asterisk boxes today to the latest cvs (up from 5/3/04).

These two boxes talk to eachother via sip, not iax.  Since the upgrade, I
get the error Failed to authenticate on INVITE trying to make calls to/from
either box.  Removing the secret from each box's sip config seems to work but
is utterly braindead.

Has anyone seen this?

- Eric
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[Asterisk-Users] Failed to authenticate on INVITE

2004-05-14 Thread Echchelh Zouhair

Hi,

I need to inteconnection to  VoIP Provider but I have this error message
when i try to dial external number :

 -- Executing SetCallerID(SIP/491-1f64, x  x ) in
new stack
-- Executing Dial(SIP/491-1f64, SIP/[EMAIL PROTECTED]|30|r)
in new stack
-- Called [EMAIL PROTECTED]

May 13 18:30:16 NOTICE[294931]: chan_sip.c:5013 handle_response: Failed to
authenticate on INVITE to 'x
SIP/[EMAIL PROTECTED];tag=as3f82df08'

Any help are welcome.
Sorry for the first message;
Zouhair Echchelh.
OPTION-SERVICE.FR


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