I am using asterisk ( ver 13.10.0.0) as a very simple PBX. I have two trunks I want it to register with and make/accept sip calls to/from.
My config works for each single trunk but when I combined them only 1 trunk shows in the ‘show sip peers’ output. BTW I have inherited this setup so am still learning about asterisk and sip.
The switches the PBX is registering to shows each show a succesful registration.
ie switch 1 & switch 2 both think PBX is registered succesfully.
Calls from a switch that thinks PBX is registered with but PBX disagrees fail.
My SIP.conf is below:-
Starting with a simple question, should this work with two trunks okay ?or am i chasing a white elephant ?
Assuming it should work, any suggestions what is wrong ?
SIP.conf
; *****************************************************
; ** General Section - thing in here APPLY GLOBALLY! **
; *****************************************************
; --------------------------------------------------
; – stage 1 goal –
; – –
; – we want the phones to bind to 5060 –
; – we want The Shout server to bind to 5061 –
; --------------------------------------------------
[general]
context=public
; ---------------------------------------------------------
; – listen for fixed phone SIP connections on port 5060 –
; ---------------------------------------------------------
udpbindport=5060
bindaddr=10.92.160.5
svrlookup=yes
#include “/etc/asterisk/sip.extensions.conf”
; *****************************************
; ** end of GENERAL section **
; *****************************************
; --------------------------------------------------------------
; – now we must bind port 5061 for TLS on the same Interface –
; --------------------------------------------------------------
;[authentication]
tlsenable=yes ; Enable server for incoming TLS (secure) connections (default is no)
tlsbindaddr=10.154.160.5:5061
srvlookup=yes
defaultexpiry=300
tlscertfile=/etc/asterisk/keys/asterisk.pem ; Certificate chain (.pem format only) to use for TLS connections
tlsprivatekey=/etc/asterisk/keys/asterisk_private.pem ; Private key file (.pem format only) for TLS connections.
tlscafile=/etc/asterisk/keys/calist.pem
tlsdontverifyserver=yes
tlscipher=ECDHE-ECDSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:
@STRENGTH
tlsclientmethod=tlsv1.2
; *********************************
; ** RTP SECTION **
; *********************************
rtptimeout=30 ; Terminate call if 30 seconds elapse with no RTP
;
; ********************************
; ** OUTBOUND SIP REGISTRATIONS **
; ********************************
register => tls://pbx:pbx@10.92.82.2:5061 ;Switch 1
register => tls://pbx:pbx@10.92.82.3:5061 ;SWitch 2
; *******************************************************
; ** INBOUND SIP PEER SWITCH 1 DEFINITION **
; *******************************************************
[ShoutGW]
nat=yes
type=friend
host=10.92.82.2
port=5061
defaultuser=pbx
secret=pbx
context=public
insecure=invite,port
transport=tls
disallow=all
allow=alaw,ulaw
encryption=yes
directmedia=no
tlsdontverifyserver=yes
; *******************************************************
; ** INBOUND SIP PEER REF B SHOUT SWITCH 2 DEFINITION **
; *******************************************************
[ShoutGW2]
nat=yes
type=friend
host=10.92.82.3
port=5061
defaultuser=pbx
secret=pbx
context=public
insecure=invite,port
transport=tls
disallow=all
allow=alaw,ulaw
encryption=yes
directmedia=no
tlsdontverifyserver=yes