Sonny Rajagopalan
2015-Mar-25 17:58 UTC
[asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling
Hello, I have had numerous issues with PJSIP outbound calling in Asterisk 13.1.0 and SIP.US SIP trunk. My Asterisk server is on EC2 and I have opened up the appropriate ports. The SIP clients can be anywhere on the Internet, including behind NATs. I am able to get to my Asterisk server's internal extensions via the DID (and appropriate dialplans) but I am not able to make outbound calls to the PSTN from my (internal) extensions. I have the appropriate dialplans and I know the Asterisk server is getting in touch with the SIP.US server (see http://lists.digium.com/pipermail/asterisk-users/2015-March/286176.html which is the error I get). My question is, does anybody have a working pjsip.conf with SIP.US I could use? It has to be pjsip.conf (and not the wizard based configuration introduced in 13.2.0). Do I need to set up an outbound_proxy for SIP.US? Any help is deeply appreciated. Thank you! Alternately, could you help me with my config (a copy is below, changed some sensitive fields for obvious reasons)? I have configured my trunks in the following manner (based on https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Configuration+Examples, and other pages on the same wiki, but there are small changes between them which confused the heck out of me): [transport-udp] type=transport protocol=udp bind=0.0.0.0 local_net=172.31.32.0/20 local_net=192.168.1.0/24 external_media_address=aa.bb.cc.dd ; replaced real public IP address external_signaling_address=aa.bb.cc.dd ; replaced real public IP address [sonnyGW1] type=registration transport=transport-udp outbound_auth=sonnyGW1_auth server_uri=sip:registrar at gw1.sip.us ; no registrar@ in URI client_uri=sip:sonny at gw1.sip.us contact_user=16175551212 ; replaced real DID retry_interval=60 forbidden_retry_interval=600 expiration=3600 [sonnyGW1_auth] type=auth auth_type=userpass password=********** username=sonny ;realm=65.254.44.194 ;realm=gw1.sip.us [sonnyGW1] type=aor contact=sip:sonnyGW1 at 65.254.44.194:5060 ; tried also no username in URI [sonnyGW1] type=endpoint transport=transport-udp context=fromgw allow=!all,ulaw outbound_auth=sonnyGW1_auth aors=sonnyGW1 from_domain=gw1.sip.us [sonnyGW1] type=identify endpoint=sonnyGW1 match=65.254.44.194 ;; All endpoints for internal extensions follow -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150325/39c93196/attachment.html>
Apparently Analagous Threads
- PJSIP/Asterisk 13.1.0 incoming call via DID: No matching endpoint found
- Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
- Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
- Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
- Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.