Sonny Rajagopalan
2015-Mar-15 16:34 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.
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. 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/2c44a5f5/attachment.html>
George Joseph
2015-Mar-15 19:25 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 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/51a9893f/attachment.html>
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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Sonny Rajagopalan
2015-Mar-24 20:09 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.
Hi George,
Well, as it turns out the removal of "realm" in sonnyGW1_auth above
does
not remove the issue. I still see the issue. I did not see the issue
earlier likely due to the CLI logging command mixup which I have now solved
using a wireshark trace (CLI was just too verbose). I see the 407
authentication required still, and the following pattern just repeats at
the Asterisk server (which is connected to the SIP trunk at 65.254.44.194)
because the SIP trunk needs it to complete the outbound call, but the
Asterisk server doesn't ever send it even after the 407 from the SIP trunk:
Wireshark trace of failed outbound call:
217274 5915.986472000 sonnysMachine 65.254.44.194 SIP/SDP
1227 Request: INVITE sip:16175551212 at 65.254.44.194:5060 |
217280 5916.059148000 65.254.44.194 sonnysMachine SIP
385 Status: 100 Trying |
217282 5916.059909000 65.254.44.194 sonnysMachine SIP
582 Status: 407 Proxy Authentication Required |
217285 5916.060227000 sonnysMachine 65.254.44.194 SIP
425 Request: ACK sip:16175551212 at 65.254.44.194:5060 |
...
(repeats ad infinitum)
When I look at the challenge in 407 Proxy Authentication Required from the
server, I see that the realm is 65.254.44.194 (gw1.sip.us), but the
appropriate Authorization (sent in the trunk registration, for example) is
never sent back from the Asterisk server. Here's what the SIP trunk
actually says (407 Auth required message; the nonce was changed by me):
Wireshark detail of 407 Proxy Authentication Required packet from SIP trunk:
Proxy-Authenticate: Digest realm="65.254.44.194",
nonce="BLUBBERbbb/e495019b-83b4-491c-8f33-3e238a3c6af2",
qop="auth"
Authentication Scheme: Digest
Realm: "65.254.44.194"
Nonce Value:
"BLUBBERbbb/e495019b-83b4-491c-8f33-3e238a3c6af2"
QOP: "auth"
And here's how the SIP trunk registration works (correctly); note the
bigger REGISTER message in the 3rd line pertaining to the registration at
65.254.44.194, it pertains to the additional 274 bytes of authentication
information:
Wireshark detail of successful SIP trunk registration:
12634 230.390420000 sonnysMachine 65.254.44.194 SIP
543 Request: REGISTER sip:gw1.sip.us (fetch bindings) |
12635 230.461572000 65.254.44.194 sonnysMachine SIP
560 Status: 401 Unauthorized (0 bindings) |
12637 230.462041000 sonnysMachine 65.254.44.194 SIP
815 Request: REGISTER sip:gw1.sip.us (fetch bindings) |
12639 230.535100000 65.254.44.194 sonnysMachine SIP
486 Status: 200 OK (0 bindings) |
Any help is deeply appreciated.
Has anyone successfully done SIP trunk registration with PJSIP in Asterisk
13.1.0?
On Sun, Mar 15, 2015 at 12:34 PM, 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.
>
> 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
>>
>
>
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