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 <[email protected] > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > [email protected]> 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:[email protected] >> 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
