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@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@from-internal:2] Dial("PJSIP/sonny-00000031", "PJSIP/12025551212@sonnyGW1") in new stack -- Called PJSIP/12025551212@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.jos...@fairview5.com> wrote: > On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < > sonny.rajagopa...@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.jos...@fairview5.com> wrote: >> >>> >>> >>> On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < >>> sonny.rajagopa...@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:so...@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