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

[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 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
>>>
>>>
>>
>> --
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>
>
>
> --
>
> 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
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
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