Thanks michael for all the help. I¹ve seem to have fixed this by adding the public IP address in my sip profile file under the ext-rtp-ip and ext-sip-ip param. Now I can see the DTMF tones come in while capturing. Sean On 3/30/10 5:29 PM, "Michael Collins" <msc@freeswitch.org> wrote:
On Tue, Mar 30, 2010 at 5:09 PM, Sean Holt <sean@obscuradigital.com> wrote:
So it appears that Bandwidth requires the audio to be routed on the external public IP address, but since my freeswitch pbx box is behind a NAT the audio is trying to route to a private IP. The tech at bandwidth says that won¹t work.
Not sure how to correct this?
Thanks Sean
I strongly recommend you read up on NAT: http://wiki.freeswitch.org/wiki/Nat
and
http://wiki.freeswitch.org/wiki/NAT_Traversal
and
http://wiki.freeswitch.org/wiki/Auto_NAT
You need to set up your external profile properly, which means you need to understand your network and your NAT device. Did you turn off any SIP ALG's in your device? Do you have a static public IP address? Those are important questions to answer before you continue.
-MC
On 3/30/10 4:11 PM, "Michael Collins" <msc@freeswitch.org <http://msc@freeswitch.org> > wrote:
On Tue, Mar 30, 2010 at 3:56 PM, Sean Holt <sean@obscuradigital.com <http://sean@obscuradigital.com> > wrote:
Unfortunately I don¹t see any reference to DTMF coming from outside source.
It shows it¹s using 711
Thanks Sean
What about listening to the call? Do you hear any DTMFs in the audio stream? If not then it looks like the far end isn't sending DTMFs which means you will need to talk to your provider to continue troubleshooting.
-MC
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