[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 Joseph george.joseph at fairview5.com
Sun Mar 15 15:00:54 CDT 2015 

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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/37023db5/attachment.html>

More information about the asterisk-users mailing list