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

[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 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"  >;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

[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 wrote:

> Please help me..
>
> Best Regards,
> *Jayesh Labade*
> e-mail: jayesh.lab...@gmail.com
>
>
>
> On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 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" > >;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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> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   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
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Re: [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@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  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 wrote:
>
>> Please help me..
>>
>> Best Regards,
>> *Jayesh Labade*
>> e-mail: jayesh.lab...@gmail.com
>>
>>
>>
>> On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 
>> 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" >> >;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
>>>
>>>
>>
>> --
>> _
>> -- 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
>
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> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [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 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@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  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 wrote:
>>
>>> Please help me..
>>>
>>> Best Regards,
>>> *Jayesh Labade*
>>> e-mail: jayesh.lab...@gmail.com
>>>
>>>
>>>
>>> On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 
>>> 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" >>> >;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


>>>
>>> --
>>> _
>>> -- 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
>>
>
>
> --
> _
> -- 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

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

[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"
;tag=as417a5527'


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



On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati  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 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@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 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 
>>> wrote:
>>>
 Please help me..

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



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade >>> > 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"  >;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
>
>

 --
 _
 -- 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
>>>
>>
>>
>> --
>> _
>> -- 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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New 

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

[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 



On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 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"
;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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Try:
register => test02:test02@192.168.1.55/s

sean



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