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 <[email protected]> wrote: > On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < > [email protected] > <javascript:_e(%7B%7D,'cvml','[email protected]');>> 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@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. >> > > 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 < >> [email protected] >> <javascript:_e(%7B%7D,'cvml','[email protected]');>> wrote: >> >>> On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < >>> [email protected] >>> <javascript:_e(%7B%7D,'cvml','[email protected]');>> 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 < >>>> [email protected] >>>> <javascript:_e(%7B%7D,'cvml','[email protected]');>> wrote: >>>> >>>>> >>>>> >>>>> On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < >>>>> [email protected] >>>>> <javascript:_e(%7B%7D,'cvml','[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] >>>>>> <javascript:_e(%7B%7D,'cvml','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 >>>> >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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
