[asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling
sonny.rajagopalan at gmail.com
Wed Mar 25 12:58:22 CDT 2015
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
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.
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
and other pages on the same wiki, but there are small changes between them
which confused the heck out of me):
external_media_address=aa.bb.cc.dd ; replaced real public IP address
external_signaling_address=aa.bb.cc.dd ; replaced real public IP address
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
contact=sip:sonnyGW1 at 220.127.116.11:5060 ; tried also no username in URI
;; All endpoints for internal extensions follow
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