[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.

Sonny Rajagopalan sonny.rajagopalan at gmail.com
Sun Mar 15 14:33:22 CDT 2015


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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/beda1bae/attachment.html>


More information about the asterisk-users mailing list