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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/beda1bae/attachment.html>
George Joseph
2015-Mar-15 20:00 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.
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>
Sonny Rajagopalan
2015-Mar-15 20:17 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.
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 >> > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/f096bdfc/attachment.html>
Sonny Rajagopalan
2015-Mar-15 23:56 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.
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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/23f90430/attachment-0001.html>
George Joseph
2015-Mar-16 01:37 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.
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 -- > 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
Maybe Matching Threads
- 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 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
- 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 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
- sysvolcheck and aclcheck generates error