Capture the media between the ATA and FS. Analyze w/ Wireshark - is the ATA *really* sending RFC2833? Check out this cool video on how to do SIP debugging on your network. You won't be disappointed. http://www.viddler.com/explore/cluecon/videos/39/ -MC On Thu, Feb 17, 2011 at 12:47 PM, Charles Bujold <cjbujold@accra.ca> wrote:
Already set to RFC2833, any other ideas?
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto: freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* February-17-11 3:41 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Unable to dial IVR menu options
Make sure that you have the RFC2833 turned on in the unit.
http://wiki.freeswitch.org/wiki/Interop_List#Grandstream_HandyTone_503
-MC
On Thu, Feb 17, 2011 at 11:20 AM, Charles Bujold <cjbujold@accra.ca> wrote:
Here is the scenario, I have a telephone connected to a Grandstream HT503 fxs with a pstn connected to the fxo and a SIP connection via the Lan to a freeswitch PBX. If I dial my voicemail (located on the SIP Freeswitch) I get the IVR for the voice mail but when I dial my password it does not get the numbers (password#) I dial to hear my messages. If I try this from a SIP phone everything works and I can dial an IVR option.
It looks like if Freeswitch is not able to pickup the dial numbers of the phone once the call is started. Any suggestion on how to fix?
Thanks cjb
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