Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP trunk outbound calling


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
sonny.rajagopalan at g...
Guest





PostPosted: Wed Mar 25, 2015 12:59 pm    Post subject: [asterisk-users] PJSIP configuration for Asterisk 13.1.0/SIP Reply with quote

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@gw1.sip.us ([email]sip%3Aregistrar@gw1.sip.us[/email]) ; no registrar@ in URI
client_uri=sip:sonny@gw1.sip.us ([email]sip%3Asonny@gw1.sip.us[/email])
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@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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services