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

[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.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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>>> 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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
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