[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 11:19:47 CDT 2015
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
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150315/259dc6af/attachment.html>
More information about the asterisk-users
mailing list