[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 15:17:18 CDT 2015
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:
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>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> _____________________________________________________________________
>>>>> -- 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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>>>>>
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>>> http://www.asterisk.org/hello
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>>>
>>>
>>> --
>>> _____________________________________________________________________
>>> -- 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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>>
>>
>> --
>> _____________________________________________________________________
>> -- 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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