Hi all,
I am new in asterisk,i tried to find a solution to my problem in the forum but unfortunately i couldn’t.
I would appreciate your help in this.
Problem:
I have configured SIP trunk in my asterisk (version 1.8.32.2) with an Avaya PBX.
Avaya supports only SIP trunking over TCP, so i configured the sip.conf of my asterisk like below:
[i][general]
bindport=5060
tcpenable=yes
transport=tcp
[Avaya]
type=friend
transport=tcp[/i]
Output of “sip show settings” command below:
[i]Global Settings:
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: 0.0.0.0:5060
Default Settings:
Allowed transports: TCP
Outbound transport: TCP[/i]
The sip signaling of incoming calls to my asterisk works fine over TCP.
But when i try outbound call (e.g. bridge transfer), the INVITE message my asterisk sends to Avaya is over UDP? and of course it fails.
INVITE sip:20000@172.x.x.20 SIP/2.0
192.168.x.x:5060 172.x.x.20:5060 xVia: SIP/2.0/UDP 192.168.x.x:5060;branch=z9hG4bK6a71196d;rport
Could you please tell me what exactly is missing in my configurations and the INVITE message that my asterisk send to Avaya goes through UDP?
Thanks!