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

Wed Jan 4 05:14:20 CST 2012

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 at anonymous.invalid:192>;tag=as417a5527'

Best Regards,
*Jayesh Labade*
e-mail: jayesh.labade at gmail.com

On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati <virbhati at 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.labade at gmail.com>wrote:
>
>> 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 at 192.168.1.55
>>
>>
>> [test02]
>> type=peer
>> nat=no
>> canreinvite=no
>> host=192.168.1.55
>> ;realm=test02 at 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 at 192.168.1.55. And if tried to register same account in
>> asterisk trunk i got F=sip:test02 at 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.labade at gmail.com
>>
>>
>>
>> On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati <virbhati at 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.labade at gmail.com>wrote:
>>>
>>>> Please help me..
>>>>
>>>> Best Regards,
>>>> *Jayesh Labade*
>>>> e-mail: jayesh.labade at gmail.com
>>>>
>>>>
>>>>
>>>> On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade <jayesh.labade at 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 at 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.labade at gmail.com
>>>>>
>>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
>>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>>               http://www.asterisk.org/hello
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>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>>
>>> --
>>>
>>> Thanks and regards
>>>
>>>  Virendra Bhati
>>> +91-8885268942
>>> Software Engineer
>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>               http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>               http://www.asterisk.org/hello
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-8885268942
> Software Engineer
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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