I have the following setup:
UA1 (SPA2000) -- Nat1 -- Asterisk (public internet) -- Nat 1 -- UA2 (X-Lite)
Relevant parts of sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
externip = 60.234.100.100 ;External IP address
localnet = 192.168.1.0/255.255.255.0 ;Local network address
allow=all
[1590]
username=1590
type=friend
secret=secret
qualify=no
port=5060
nat=yes
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=test
canreinvite=no
allow=all
[1593]
username=1593
type=friend
secret=secret
qualify=no
port=5060
nat=yes
[EMAIL PROTECTED]
host=dynamic
dtmfmode=rfc2833
context=test
canreinvite=no
allow=all
I have enabled rtp debugging and notice that Asterisk is receiving no rtp
traffic. When I call from either UA to voicemail for example I see RTP
traffic
e.g. call from 1590
Got RTP packet from 60.234.200.200:38510 (type 0, seq 1245, ts 207620, len
160)
Sent RTP packet to 60.234.200.200:38510 (type 0, seq 61963, ts 34880, len
160)
e.g. call from 1593
Got RTP packet from 60.234.200.200:16470 (type 0, seq 892, ts 316685167, len
240)
Sent RTP packet to 60.234.200.200:16470 (type 0, seq 1156, ts 15360, len
160)
I thought that with canreinvite=no all audio would go through Asterisk. What
have I missed?
Asterisk 1.2.13
Fedora Core 5
Regards
Cameron
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