I ran into this before to where the Adtran Router is interfering with sip traffic when I have a Pbx behind it with port forwading - 5060 and RTP port forward to the phone system. Does anyone know what I can do to tell the traffic to leave all sip and rtp traffic alone, just port forward it?
I have sip alg turn off already.
This happen to me on asterisk phone system and NEC phone system.
Did you try "no ip sip" in global config mode?
How is it interfering? What do you want to happen and what is actually happening?
Yes, I tried that and all the call between the vpn have no audio. Here is what I’m geeting from suppor
The no matching peer means that there’s no trunk set up with the senders IP address. That trunk exists from what I can see so the problem is probably network related. If he can replicate the issue, have him do a packet capture on the call that fails.
tcpdump -s 0 -w /tmp/tcpdump.cap
then replicate the issue, after the test call is rejected, hit ctrl+c to stop the tcpdump program
send me the /tmp/tcpdump.cap file. I’ve seen a case in the past where a user’s network was shaping the sip traffing and putter the gateways IP address in the IP Header field instead of the sip providers WAN address that the packet originated from. I suspect something similar may be happening. Also send me the verbose logs in /var/log/asterisk/full
When you disable the SIP handling in the 3120, then the IP address within the SIP packet for RTP won't match the source address going through NAT. You'll need to set it up as NAT=YES on the Asterisk platform in this case.
If the 3120 is handling SIP fixup, then Asterisk will want to be set up as NAT=NO.
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