All, I’ve looked quite a bit and can’t seem to find an answer to this.
We are running Asterisk 1.2.20 and have SIP DID’s point to us for incoming calls.
Extension 1000 is setup to play an announcement and then ask the user to enter a extension. If dialed from the inside on an IP phone, all is fine… The system knows that you dialed an extension.
If calling in through the DID (which points to extension 1000), the announcements play, but nothing can be dialed. It’s just ignored.
I’m not quite sure what the problem is, and have tried many things to no avail… in SIP.conf:
- dtmfmode=inband (I’ve tried all of the modes too)
Our firewall is forwarding all incoming traffic to the asterisk server and port 5060 as well as ports 10,000-12,000 are open (including UDP)
Anyone have any ideas?
Did you check about the NAT problem? Whether your call is automatically hangup after a few second?
Please make a test case of this: Instead of incomming calls is route to ivr, we reset it to a local phone. Dial to that DID to check whether outside caller can talk and hear with local phone in both 2 ways. Please check and let me know the result
Duc Viet To Ho
Thanks for your response!
You were correct… I cannot hear voice 2 ways, so the problem would seem to be the firewall… We actually did some packet sniffing and do not see the udp packets coming in through firewall…
At least we know what the problem is now…
As you told “Our firewall is forwarding all incoming traffic to the asterisk server and port 5060 as well as ports 10,000-12,000 are open (including UDP)”, it seems that your firewall is ok. Please check your system and client NAT status. I think that this might caused because of 1 way NAT issue
Duc Viet To Ho