Thanks Brian. Sorry, should have done a full sip trace before, but here is one now: <http://pastebin.freeswitch.org/12632> Calling an IVR dialplan: http://pastebin.freeswitch.org/12634 Calling from one extn to another. http://pastebin.freeswitch.org/12633 (With this one, the source/calling softphone gets a message on it saying put on hold by the other user - not sure if that helps.) For what it's worth, at a couple of points when I was running the trace I was pressing keys to generate dtmf, and nothing changed on the screen - no activity at all. Also, I've been able to remote desktop into a computer on another network, and install x-lite and it can connect to our internal server and works fine, but it can't do dtmf on the EC2 server either (so it's definitely a problem on the server end somehow, not my local network's NAT.) Cheers, Fraser On Mon, Apr 5, 2010 at 5:21 PM, Brian West <brian@freeswitch.org> wrote:
Most likely its in the RTP stream as RFC2833 which is the reason you're not getting anything plus I need a FULL sip trace not this abbreviated trace.
/b
On Apr 5, 2010, at 11:12 AM, Fraser Redmond wrote:
I've taken another stab at this one way audio problem today.
I've run a wireshark capture, and looking at the RTP analysis it only has the down-stream, it doesn't record anything being sent upstream at all.
Below is the SIP graph, which shows RTP coming down, but none going up. But I don't know enough about SIP to know whether something is missing.
Any suggestions of what I should try now?
Would the dtmf's be sent in the sip packets, or in the rtp?
To preempt the easy answers and save some time: - ports are open on EC2 config, - iptables turned off for the test, - RTP port range uncommented in switch.conf.xml, - softphone stun was set to stun.freeswitch.org
Cheers, Fraser
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