Capture the media between the ATA and FS. Analyze w/ Wireshark - is the ATA *really* sending RFC2833?<div><br></div><div>Check out this cool video on how to do SIP debugging on your network. You won't be disappointed.</div>
<div><br></div><div><a href="http://www.viddler.com/explore/cluecon/videos/39/">http://www.viddler.com/explore/cluecon/videos/39/</a></div><div><br></div><div>-MC<br><br><div class="gmail_quote">On Thu, Feb 17, 2011 at 12:47 PM, Charles Bujold <span dir="ltr"><<a href="mailto:cjbujold@accra.ca">cjbujold@accra.ca</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div lang="EN-CA" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D">Already set to RFC2833, any other ideas?</span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;color:#1F497D"> </span></p><p class="MsoNormal"><b><span lang="EN-US" style="font-size:10.0pt">From:</span></b><span lang="EN-US" style="font-size:10.0pt"> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Michael Collins<br>
<b>Sent:</b> February-17-11 3:41 PM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] Unable to dial IVR menu options</span></p><div><div></div><div class="h5"><p class="MsoNormal"> </p><p class="MsoNormal">
Make sure that you have the RFC2833 turned on in the unit. </p><div><p class="MsoNormal"><a href="http://wiki.freeswitch.org/wiki/Interop_List#Grandstream_HandyTone_503" target="_blank">http://wiki.freeswitch.org/wiki/Interop_List#Grandstream_HandyTone_503</a></p>
</div><div><p class="MsoNormal"> </p></div><div><p class="MsoNormal" style="margin-bottom:12.0pt">-MC</p><div><p class="MsoNormal">On Thu, Feb 17, 2011 at 11:20 AM, Charles Bujold <<a href="mailto:cjbujold@accra.ca" target="_blank">cjbujold@accra.ca</a>> wrote:</p>
<div><div><p class="MsoNormal" style="line-height:16.8pt;background:#E7EAEF"><span style="font-size:9.5pt;color:black">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.<br>
<br>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?<br><br>Thanks<br>cjb</span></p><p class="MsoNormal"> </p></div></div><p class="MsoNormal" style="margin-bottom:12.0pt">
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