Unfortunately I don¹t see any reference to DTMF coming from outside source. It shows it¹s using 711 Thanks Sean On 3/30/10 12:50 PM, "Phillip Jones" <pjintheusa@gmail.com> wrote:
Use wireshark to see (and playback) the DTMF coming down the line.
For reliable inband I believe you need to be using 711 - so that is worth checking too.
On Tue, Mar 30, 2010 at 3:37 PM, Sean Holt <sean@obscuradigital.com> wrote:
Yea still getting no love. Had bandwidth turn on inband but still no success.
On 3/30/10 12:10 PM, "Michael Collins" <msc@freeswitch.org <http://msc@freeswitch.org> > wrote:
The start_dtmf app simply turns on listening in-band for DTMFs, so this should be just fine. Have bw.com <http://bw.com> <http://bw.com> enable in-band DTMFs and give it a test drive.
-MC
On Tue, Mar 30, 2010 at 11:01 AM, Sean Holt <sean@obscuradigital.com <http://sean@obscuradigital.com> > wrote:
Hello,
I¹ve posted on the question on this topic before but I¹ll try again with more detail.
I have this svn version FreeSWITCH Version 1.0.trunk (16952) SIP provider: Bandwidth.com
I¹m assuming that DTMF rc2833 is the default, but I can¹t get my ivr to recognize the incoming call tones. Bandwidth can set up either inband or RC2833. They require these codecs, G.711Ulaw and G.729A.
The tech at Bandwidth wanted me to setup inband to test if that works better, but wasn¹t sure where to set it up. I added this to my dialplan
<extension name="main_number_23" continue="true"> <condition field="destination_number" expression="^3000$"> <action application="answer"/> <action application="start_dtmf"/> <action application="sleep" data="1000"/> <action application="conference" data="conference_1@conference_1"/> </condition> </extension>
Should I add it to my sip trunk?
Not sure how to troubleshoot this issue
Thanks Sean
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