thr3ads.net - asterisk users - [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge. [Mar 2015]

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Sonny Rajagopalan

2015-Mar-15 19:33 UTC

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

Yes, I think the dial does get executed (sonny calling outbound
202-555-1212):

core set verbose 3
Console verbose was OFF and is now 3.
    -- Executing [912025551212 at from-internal:1]
Log("PJSIP/sonny-00000031",
"NOTICE, Dialing out from "" <sonny> to 12025551212 through
fromgw") in new
stack
[Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @
from-internal:  Dialing out from "" <sonny> to 12025551212
through fromgw
    -- Executing [912025551212 at from-internal:2]
Dial("PJSIP/sonny-00000031", "PJSIP/12025551212 at
sonnyGW1") in new stack
    -- Called PJSIP/12025551212 at sonnyGW1

the number 202-555-1212 does not ring.

at hangup on caller (sonny):

  == Spawn extension (from-internal, 912025551212, 2) exited non-zero on
'PJSIP/sonny-00000031'

On Sun, Mar 15, 2015 at 3:25 PM, George Joseph <george.joseph at
fairview5.com>
wrote:
> On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> That was the issue, thanks. I now am able to get the caller ringing on
an
>> outbound call, but an external phone number (E164) I am dialing does
not
>> ring.
>>
>
> Any error messages?  If you set 'core set verbose 3' and try it,
does the
> Dial get executed?
>
>
>
>>
>> On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <
>> george.joseph at fairview5.com> wrote:
>>
>>>
>>>
>>> On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan <
>>> sonny.rajagopalan at gmail.com> wrote:
>>>
>>>> I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2.
My basic
>>>> configuration works, and I am connected to a SIP trunk using
SIP.US,
>>>> and have set up my inbound calling which works correctly (when
I call my
>>>> PBX DID, the call does come into my PBX network).
>>>>
>>>> The issue is that I am not able to make outbound calls, because
the
>>>> call fails with the error:
>>>>
>>>> res_pjsip_outbound_authenticator_digest.c:125
>>>> digest_create_request_with_auth: Unable to create request with
auth.No auth
>>>> credentials for any realms in challenge.
>>>>
>>>> CLI> pjsip show endpoint sonnyGW1
>>>>
>>>> ...
>>>>
========================================================================================>>>>
>>>>  Endpoint:  sonnyGW1                                        Not
in use
>>>>    0 of inf
>>>>     OutAuth:  sonnyGW1_auth/sonny
>>>>         Aor:  sonnyGW1                                      0
>>>>       Contact:  sonnyGW1/sip:65.254.44.194:5060            
Unknown
>>>>             nan
>>>>   Transport:  transport-udp             udp      0      0 
0.0.0.0:5060
>>>>    Identify:  sonnyGW1/sonnyGW1
>>>>         Match: 65.254.44.194/32
>>>>
>>>> My pjsip.conf is as below
>>>>
>>>> [sonnyGW1]
>>>> type=registration
>>>> transport=transport-udp
>>>> outbound_auth=sonnyGW1_auth
>>>> server_uri=sip:gw1.sip.us
>>>> client_uri=sip:sonny at gw1.sip.us
>>>> contact_user=sonny
>>>> retry_interval=60
>>>> forbidden_retry_interval=600
>>>> expiration=3600
>>>>
>>>> [sonnyGW1_auth]
>>>> type=auth
>>>> auth_type=userpass
>>>> password=somepassword
>>>> username=sonny
>>>> realm=gw1.sip.us
>>>>
>>>
>>> You probably need to remove the 'realm' line so that it
will match any
>>> realm in the challenge.
>>>
>>>
>>>>
>>>> [sonnyGW1]
>>>> type=aor
>>>> contact=sip:65.254.44.194:5060
>>>>
>>>> [sonnyGW1]
>>>> type=endpoint
>>>> transport=transport-udp
>>>> context=gateway1
>>>> allow=!all,ulaw
>>>> outbound_auth=sonnyGW1_auth
>>>> aors=sonnyGW1
>>>>
>>>> [sonnyGW1]
>>>> type=identify
>>>> endpoint=sonnyGW1
>>>> match=65.254.44.194
>>>>
>>>> My extensions.conf stub for the appropriate section looks like
this
>>>> (from
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels)
>>>> :
>>>>
>>>> exten => _9XXXX.,1,Log(NOTICE, Dialing out from
${CALLERID(all)} to
>>>> ${EXTEN:1} through gateway1)
>>>> exten => _9XXXX.,n,Dial(PJSIP/${EXTEN:1}@sonnyGW1)
>>>> ; Have also tried
>>>> ; exten =>
_9XXXX.,n,Dial(PJSIP/sip:${EXTEN:1}@65.254.44.194:5060)
>>>> exten => _9XXXX.,n,Playtones(congestion)
>>>> exten => _9XXXX.,n,Hangup()
>>>>
>>>> I do know that this code is being executed as I see the log in
the
>>>> first line above.
>>>>
>>>> Have I correctly set up authentication for outbound calling?
>>>>
>>>> Any help appreciated. Thanks!
>>>>
>>>> --
>>>>
_____________________________________________________________________
>>>> -- 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
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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
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George Joseph

2015-Mar-15 20:00 UTC

head link

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Yes, I think the dial does get executed (sonny calling outbound
> 202-555-1212):
>
> core set verbose 3
> Console verbose was OFF and is now 3.
>     -- Executing [912025551212 at from-internal:1]
> Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from
"" <sonny> to
> 12025551212 through fromgw") in new stack
> [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @
> from-internal:  Dialing out from "" <sonny> to 12025551212
through fromgw
>     -- Executing [912025551212 at from-internal:2]
> Dial("PJSIP/sonny-00000031", "PJSIP/12025551212 at
sonnyGW1") in new stack
>     -- Called PJSIP/12025551212 at sonnyGW1
>
> the number 202-555-1212 does not ring.
>
You're probably going to have to turn on debug for the pjsip endpoint with
'pjsip set logger host <server>' and look at the actual outbound
INVITE and
any response.

>
> at hangup on caller (sonny):
>
>   == Spawn extension (from-internal, 912025551212, 2) exited non-zero on
> 'PJSIP/sonny-00000031'
>
> On Sun, Mar 15, 2015 at 3:25 PM, George Joseph <
> george.joseph at fairview5.com> wrote:
>
>> On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan <
>> sonny.rajagopalan at gmail.com> wrote:
>>
>>> That was the issue, thanks. I now am able to get the caller ringing
on
>>> an outbound call, but an external phone number (E164) I am dialing
does not
>>> ring.
>>>
>>
>> Any error messages?  If you set 'core set verbose 3' and try
it, does the
>> Dial get executed?
>>
>>
>>
>>>
>>> On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <
>>> george.joseph at fairview5.com> wrote:
>>>
>>>>
>>>>
>>>> On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan <
>>>> sonny.rajagopalan at gmail.com> wrote:
>>>>
>>>>> I have setup my Asterisk 13.1.0 server with PJSIP on
AWS/EC2. My basic
>>>>> configuration works, and I am connected to a SIP trunk
using SIP.US,
>>>>> and have set up my inbound calling which works correctly
(when I call my
>>>>> PBX DID, the call does come into my PBX network).
>>>>>
>>>>> The issue is that I am not able to make outbound calls,
because the
>>>>> call fails with the error:
>>>>>
>>>>> res_pjsip_outbound_authenticator_digest.c:125
>>>>> digest_create_request_with_auth: Unable to create request
with auth.No auth
>>>>> credentials for any realms in challenge.
>>>>>
>>>>> CLI> pjsip show endpoint sonnyGW1
>>>>>
>>>>> ...
>>>>>
========================================================================================>>>>>
>>>>>  Endpoint:  sonnyGW1                                       
Not in use
>>>>>    0 of inf
>>>>>     OutAuth:  sonnyGW1_auth/sonny
>>>>>         Aor:  sonnyGW1                                     
0
>>>>>       Contact:  sonnyGW1/sip:65.254.44.194:5060            
Unknown
>>>>>             nan
>>>>>   Transport:  transport-udp             udp      0      0
>>>>> 0.0.0.0:5060
>>>>>    Identify:  sonnyGW1/sonnyGW1
>>>>>         Match: 65.254.44.194/32
>>>>>
>>>>> My pjsip.conf is as below
>>>>>
>>>>> [sonnyGW1]
>>>>> type=registration
>>>>> transport=transport-udp
>>>>> outbound_auth=sonnyGW1_auth
>>>>> server_uri=sip:gw1.sip.us
>>>>> client_uri=sip:sonny at gw1.sip.us
>>>>> contact_user=sonny
>>>>> retry_interval=60
>>>>> forbidden_retry_interval=600
>>>>> expiration=3600
>>>>>
>>>>> [sonnyGW1_auth]
>>>>> type=auth
>>>>> auth_type=userpass
>>>>> password=somepassword
>>>>> username=sonny
>>>>> realm=gw1.sip.us
>>>>>
>>>>
>>>> You probably need to remove the 'realm' line so that it
will match any
>>>> realm in the challenge.
>>>>
>>>>
>>>>>
>>>>> [sonnyGW1]
>>>>> type=aor
>>>>> contact=sip:65.254.44.194:5060
>>>>>
>>>>> [sonnyGW1]
>>>>> type=endpoint
>>>>> transport=transport-udp
>>>>> context=gateway1
>>>>> allow=!all,ulaw
>>>>> outbound_auth=sonnyGW1_auth
>>>>> aors=sonnyGW1
>>>>>
>>>>> [sonnyGW1]
>>>>> type=identify
>>>>> endpoint=sonnyGW1
>>>>> match=65.254.44.194
>>>>>
>>>>> My extensions.conf stub for the appropriate section looks
like this
>>>>> (from
>>>>>
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels) :
>>>>>
>>>>> exten => _9XXXX.,1,Log(NOTICE, Dialing out from
${CALLERID(all)} to
>>>>> ${EXTEN:1} through gateway1)
>>>>> exten => _9XXXX.,n,Dial(PJSIP/${EXTEN:1}@sonnyGW1)
>>>>> ; Have also tried
>>>>> ; exten =>
_9XXXX.,n,Dial(PJSIP/sip:${EXTEN:1}@65.254.44.194:5060)
>>>>> exten => _9XXXX.,n,Playtones(congestion)
>>>>> exten => _9XXXX.,n,Hangup()
>>>>>
>>>>> I do know that this code is being executed as I see the log
in the
>>>>> first line above.
>>>>>
>>>>> Have I correctly set up authentication for outbound
calling?
>>>>>
>>>>> Any help appreciated. Thanks!
>>>>>
>>>>> --
>>>>>
_____________________________________________________________________
>>>>> -- 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
>>>>
>>>
>>>
>>> --
>>>
_____________________________________________________________________
>>> -- 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
>>
>
>
> --
> _____________________________________________________________________
> -- 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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Sonny Rajagopalan

2015-Mar-15 20:17 UTC

head link

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

I am out now, and can happily send details in a couple of hours. However, I
can give you a summary of what happens. The PBX sends an invite and I
immediately start ringing at the caller (100 trying) and the I get a 407
proxy auth required to which the server responds but clearly the sip
gateway is not happy with this.

Thank you for responding!

On Sunday, March 15, 2015, George Joseph <george.joseph at fairview5.com>
wrote:
> On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com
> <javascript:_e(%7B%7D,'cvml','sonny.rajagopalan at
gmail.com');>> wrote:
>
>> Yes, I think the dial does get executed (sonny calling outbound
>> 202-555-1212):
>>
>> core set verbose 3
>> Console verbose was OFF and is now 3.
>>     -- Executing [912025551212 at from-internal:1]
>> Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from
"" <sonny> to
>> 12025551212 through fromgw") in new stack
>> [Mar 15 19:27:06] NOTICE[16648][C-00000022]: Ext. 912025551212:1 @
>> from-internal:  Dialing out from "" <sonny> to
12025551212 through fromgw
>>     -- Executing [912025551212 at from-internal:2]
>> Dial("PJSIP/sonny-00000031", "PJSIP/12025551212 at
sonnyGW1") in new stack
>>     -- Called PJSIP/12025551212 at sonnyGW1
>>
>> the number 202-555-1212 does not ring.
>>
>
> You're probably going to have to turn on debug for the pjsip endpoint
with
> 'pjsip set logger host <server>' and look at the actual
outbound INVITE and
> any response.
>
>
>>
>> at hangup on caller (sonny):
>>
>>   == Spawn extension (from-internal, 912025551212, 2) exited non-zero
on
>> 'PJSIP/sonny-00000031'
>>
>> On Sun, Mar 15, 2015 at 3:25 PM, George Joseph <
>> george.joseph at fairview5.com
>> <javascript:_e(%7B%7D,'cvml','george.joseph at
fairview5.com');>> wrote:
>>
>>> On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan <
>>> sonny.rajagopalan at gmail.com
>>> <javascript:_e(%7B%7D,'cvml','sonny.rajagopalan at
gmail.com');>> wrote:
>>>
>>>> That was the issue, thanks. I now am able to get the caller
ringing on
>>>> an outbound call, but an external phone number (E164) I am
dialing does not
>>>> ring.
>>>>
>>>
>>> Any error messages?  If you set 'core set verbose 3' and
try it, does
>>> the Dial get executed?
>>>
>>>
>>>
>>>>
>>>> On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <
>>>> george.joseph at fairview5.com
>>>> <javascript:_e(%7B%7D,'cvml','george.joseph at
fairview5.com');>> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan <
>>>>> sonny.rajagopalan at gmail.com
>>>>>
<javascript:_e(%7B%7D,'cvml','sonny.rajagopalan at
gmail.com');>> wrote:
>>>>>
>>>>>> I have setup my Asterisk 13.1.0 server with PJSIP on
AWS/EC2. My
>>>>>> basic configuration works, and I am connected to a SIP
trunk using
>>>>>> SIP.US, and have set up my inbound calling which works
correctly
>>>>>> (when I call my PBX DID, the call does come into my PBX
network).
>>>>>>
>>>>>> The issue is that I am not able to make outbound calls,
because the
>>>>>> call fails with the error:
>>>>>>
>>>>>> res_pjsip_outbound_authenticator_digest.c:125
>>>>>> digest_create_request_with_auth: Unable to create
request with auth.No auth
>>>>>> credentials for any realms in challenge.
>>>>>>
>>>>>> CLI> pjsip show endpoint sonnyGW1
>>>>>>
>>>>>> ...
>>>>>>
========================================================================================>>>>>>
>>>>>>  Endpoint:  sonnyGW1                                   
Not in
>>>>>> use    0 of inf
>>>>>>     OutAuth:  sonnyGW1_auth/sonny
>>>>>>         Aor:  sonnyGW1                                 
0
>>>>>>       Contact:  sonnyGW1/sip:65.254.44.194:5060        
Unknown
>>>>>>               nan
>>>>>>   Transport:  transport-udp             udp      0     
0
>>>>>> 0.0.0.0:5060
>>>>>>    Identify:  sonnyGW1/sonnyGW1
>>>>>>         Match: 65.254.44.194/32
>>>>>>
>>>>>> My pjsip.conf is as below
>>>>>>
>>>>>> [sonnyGW1]
>>>>>> type=registration
>>>>>> transport=transport-udp
>>>>>> outbound_auth=sonnyGW1_auth
>>>>>> server_uri=sip:gw1.sip.us
>>>>>> client_uri=sip:sonny at gw1.sip.us
>>>>>> <javascript:_e(%7B%7D,'cvml','sip:sonny
at gw1.sip.us');>
>>>>>> contact_user=sonny
>>>>>> retry_interval=60
>>>>>> forbidden_retry_interval=600
>>>>>> expiration=3600
>>>>>>
>>>>>> [sonnyGW1_auth]
>>>>>> type=auth
>>>>>> auth_type=userpass
>>>>>> password=somepassword
>>>>>> username=sonny
>>>>>> realm=gw1.sip.us
>>>>>>
>>>>>
>>>>> You probably need to remove the 'realm' line so
that it will match any
>>>>> realm in the challenge.
>>>>>
>>>>>
>>>>>>
>>>>>> [sonnyGW1]
>>>>>> type=aor
>>>>>> contact=sip:65.254.44.194:5060
>>>>>>
>>>>>> [sonnyGW1]
>>>>>> type=endpoint
>>>>>> transport=transport-udp
>>>>>> context=gateway1
>>>>>> allow=!all,ulaw
>>>>>> outbound_auth=sonnyGW1_auth
>>>>>> aors=sonnyGW1
>>>>>>
>>>>>> [sonnyGW1]
>>>>>> type=identify
>>>>>> endpoint=sonnyGW1
>>>>>> match=65.254.44.194
>>>>>>
>>>>>> My extensions.conf stub for the appropriate section
looks like this
>>>>>> (from
>>>>>>
https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels) :
>>>>>>
>>>>>> exten => _9XXXX.,1,Log(NOTICE, Dialing out from
${CALLERID(all)} to
>>>>>> ${EXTEN:1} through gateway1)
>>>>>> exten => _9XXXX.,n,Dial(PJSIP/${EXTEN:1}@sonnyGW1)
>>>>>> ; Have also tried
>>>>>> ; exten =>
_9XXXX.,n,Dial(PJSIP/sip:${EXTEN:1}@65.254.44.194:5060)
>>>>>> exten => _9XXXX.,n,Playtones(congestion)
>>>>>> exten => _9XXXX.,n,Hangup()
>>>>>>
>>>>>> I do know that this code is being executed as I see the
log in the
>>>>>> first line above.
>>>>>>
>>>>>> Have I correctly set up authentication for outbound
calling?
>>>>>>
>>>>>> Any help appreciated. Thanks!
>>>>>>
>>>>>> --
>>>>>>
_____________________________________________________________________
>>>>>> -- 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
>>>>>
>>>>
>>>>
>>>> --
>>>>
_____________________________________________________________________
>>>> -- 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
>>>
>>
>>
>> --
>> _____________________________________________________________________
>> -- 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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Sonny Rajagopalan

2015-Mar-15 23:56 UTC

head link

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

George,

I have the detailed log below. (Resending after trimming the email to 40KB.)

The sequence below just repeats ad-nauseam. Is this a SIP trunk issue?

Thanks!

---------------------

Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 --->
INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>
Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060>
Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6753 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
in-SE: 90
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1014372762 1014372762 IN IP4 192.168.13.121
s=Asterisk
c=IN IP4 18.18.19.123
t=0 0
m=audio 11614 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[Kip-192.168.13.121*CLI> [0K<--- Received SIP response (342 bytes) from
UDP:
65.254.44.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport=5060;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2;received=18.18.19.123
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>
Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6753 INVITE
Content-Length: 0


[Kip-192.168.13.121*CLI> [0K<--- Received SIP response (539 bytes) from
UDP:
65.254.44.194:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport=5060;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2;received=18.18.19.123
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.188aCall-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6753 INVITE
Proxy-Authenticate: Digest realm="65.254.44.194",
nonce="VQYKoVUGCEmlb1riTSTWwGKGMJZeqn7uVKO5AGraoWJnidF+hUD12HhxBszB",
qop="auth"
Content-Length: 0


[Kip-192.168.13.121*CLI> [0K<--- Transmitting SIP request (382 bytes) to
UDP:65.254.44.194:5060 --->
ACK sip:12025551212 at 65.254.44.194:5060 SIP/2.0
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.188aCall-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6753 ACK
Content-Length:  0


[Kip-192.168.13.121*CLI> [0K<--- Transmitting SIP request (1186 bytes) to
UDP:65.254.44.194:5060 --->
INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>
Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060>
Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6754 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
in-SE: 90
Proxy-Authorization: Digest username="sonny",
realm="65.254.44.194",
nonce="VQYKoVUGCEmlb1riTSTWwGKGMJZeqn7uVKO5AGraoWJnidF+hUD12HhxBszB",
uri="
sip:12025551212 at 65.254.44.194:5060",
response="18623a3fec388b9b85dc8f2bcd023083",
cnonce="4616a975-1521-4db5-b4d5-53b33ffba4b6", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length:   239

v=0
o=- 1014372762 1014372762 IN IP4 192.168.13.121
s=Asterisk
c=IN IP4 18.18.19.123
t=0 0
m=audio 11614 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

[Kip-192.168.13.121*CLI> [0K<--- Received SIP response (342 bytes) from
UDP:
65.254.44.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport=5060;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076;received=18.18.19.123
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>
Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6754 INVITE
Content-Length: 0


[Kip-192.168.13.121*CLI> [0K<--- Received SIP response (539 bytes) from
UDP:
65.254.44.194:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport=5060;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076;received=18.18.19.123
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.f6bcCall-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6754 INVITE
Proxy-Authenticate: Digest realm="65.254.44.194",
nonce="VQYKoVUGCEkkNCFzVwnTNw69uvD+cH8QVKO5AGraoWJnidF+hUD12HK5Ps7A",
qop="auth"
Content-Length: 0


[Kip-192.168.13.121*CLI> [0K<--- Transmitting SIP request (382 bytes) to
UDP:65.254.44.194:5060 --->
ACK sip:12025551212 at 65.254.44.194:5060 SIP/2.0
Via: SIP/2.0/UDP 18.18.19.123:5060
;rport;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076
From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
To: <sip:12025551212 at 65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.f6bcCall-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
CSeq: 6754 ACK
Content-Length:  0
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George Joseph

2015-Mar-16 01:37 UTC

head link

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan
<sonny.rajagopalan at gmail.com> wrote:> George,
>
> I have the detailed log below. (Resending after trimming the email to
40KB.)
>
> The sequence below just repeats ad-nauseam. Is this a SIP trunk issue?
>
> Thanks!
>
I don't see anything obvious.  The registration works though, right?
You might want to compare the register auth exchange to the invite
auth exchange and see if anything differs.
> ---------------------
>
> Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 --->
> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To: <sip:12025551212 at 65.254.44.194>
> Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at
18.18.19.123:5060>
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6753 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
> UPDATE, PRACK, REGISTER, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> in-SE: 90
> Content-Type: application/sdp
> Content-Length:   239
>
> v=0
> o=- 1014372762 1014372762 IN IP4 192.168.13.121
> s=Asterisk
> c=IN IP4 18.18.19.123
> t=0 0
> m=audio 11614 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
>
> [Kip-192.168.13.121*CLI> [0K<--- Received SIP response (342 bytes)
from
> UDP:65.254.44.194:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport=5060;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2;received=18.18.19.123
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To: <sip:12025551212 at 65.254.44.194>
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6753 INVITE
> Content-Length: 0
>
>
> [Kip-192.168.13.121*CLI> [0K<--- Received SIP response (539 bytes)
from
> UDP:65.254.44.194:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport=5060;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2;received=18.18.19.123
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To:
> <sip:12025551212 at
65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.188a
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6753 INVITE
> Proxy-Authenticate: Digest realm="65.254.44.194",
>
nonce="VQYKoVUGCEmlb1riTSTWwGKGMJZeqn7uVKO5AGraoWJnidF+hUD12HhxBszB",
> qop="auth"
> Content-Length: 0
>
>
> [Kip-192.168.13.121*CLI> [0K<--- Transmitting SIP request (382 bytes)
to
> UDP:65.254.44.194:5060 --->
> ACK sip:12025551212 at 65.254.44.194:5060 SIP/2.0
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To:
> <sip:12025551212 at
65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.188a
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6753 ACK
> Content-Length:  0
>
>
> [Kip-192.168.13.121*CLI> [0K<--- Transmitting SIP request (1186
bytes) to
> UDP:65.254.44.194:5060 --->
> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To: <sip:12025551212 at 65.254.44.194>
> Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at
18.18.19.123:5060>
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6754 INVITE
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL,
> UPDATE, PRACK, REGISTER, MESSAGE, REFER
> Supported: 100rel, timer, replaces, norefersub
> Session-Expires: 1800
> in-SE: 90
> Proxy-Authorization: Digest username="sonny",
realm="65.254.44.194",
>
nonce="VQYKoVUGCEmlb1riTSTWwGKGMJZeqn7uVKO5AGraoWJnidF+hUD12HhxBszB",
> uri="sip:12025551212 at 65.254.44.194:5060",
> response="18623a3fec388b9b85dc8f2bcd023083",
> cnonce="4616a975-1521-4db5-b4d5-53b33ffba4b6", qop=auth,
nc=00000001
> Content-Type: application/sdp
> Content-Length:   239
>
> v=0
> o=- 1014372762 1014372762 IN IP4 192.168.13.121
> s=Asterisk
> c=IN IP4 18.18.19.123
> t=0 0
> m=audio 11614 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
>
> [Kip-192.168.13.121*CLI> [0K<--- Received SIP response (342 bytes)
from
> UDP:65.254.44.194:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport=5060;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076;received=18.18.19.123
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To: <sip:12025551212 at 65.254.44.194>
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6754 INVITE
> Content-Length: 0
>
>
> [Kip-192.168.13.121*CLI> [0K<--- Received SIP response (539 bytes)
from
> UDP:65.254.44.194:5060 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport=5060;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076;received=18.18.19.123
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To:
> <sip:12025551212 at
65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.f6bc
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6754 INVITE
> Proxy-Authenticate: Digest realm="65.254.44.194",
>
nonce="VQYKoVUGCEkkNCFzVwnTNw69uvD+cH8QVKO5AGraoWJnidF+hUD12HK5Ps7A",
> qop="auth"
> Content-Length: 0
>
>
> [Kip-192.168.13.121*CLI> [0K<--- Transmitting SIP request (382 bytes)
to
> UDP:65.254.44.194:5060 --->
> ACK sip:12025551212 at 65.254.44.194:5060 SIP/2.0
> Via: SIP/2.0/UDP
>
18.18.19.123:5060;rport;branch=z9hG4bKPjda9b69bb-a368-4ec0-a523-df5a920a7076
> From: <sip:sonny at
192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8
> To:
> <sip:12025551212 at
65.254.44.194>;tag=2484f1a5f06b7307c34ab1dd8d74150a.f6bc
> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3
> CSeq: 6754 ACK
> Content-Length:  0
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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asterisk users - Mar 2015 - Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

[asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.

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