when you say tones, does that mean it's inband dtmf?<br>you may need to run the start_dtmf app on the channel to engage the tone detector?<br><br>can you please file it on jira <a href="http://jira.freeswitch.org">http://jira.freeswitch.org</a> and attach<br>
a pcap and console debug log.<br><br><br><br><div class="gmail_quote">On Wed, Jan 21, 2009 at 9:53 AM, Rob Charlton <span dir="ltr"><<a href="mailto:rob.charlton@savageminds.com">rob.charlton@savageminds.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Yes, and yes. I see the DTMF events arriving when I make an incoming call.<br>
<br>
I put a breakpoint on switch_channel_dequeue_dtmf() which gets hit when<br>
I type digits after:<br>
<br>
- I originate a call to a sip extension<br>
- I receive a call from a sip extension<br>
- I receive a call from our sip trunk (from PSTN)<br>
<br>
The breakpoint doesn't get hit when I type digits after:<br>
<br>
- I originate a call via our sip trunk (to the PSTN)<br>
<br>
As regards this:<br>
<div class="Ih2E3d"> > In the latter case, I am still able to pick up DTMF digits if I use<br>
> javascript session.collectInput() - so it appears as if the DTMF tones<br>
> are being recognised by Freeswitch, but no events sent.<br>
</div>I must have been dreaming - that isn't the case at all -<br>
session.collectInput doesn't get any digits at all.<br>
<br>
We use the same SIP trunk with asterisk and that _does_ pick up DTMF<br>
tones for outbound PSTN calls.<br>
<br>
Thanks<br>
<br>
Rob<br>
<div class="Ih2E3d"><br>
<br>
Anthony Minessale wrote:<br>
> Did you try enabling all events and making a single call to make sure<br>
> you are subscribed to the right event?<br>
><br>
><br>
> On Wed, Jan 21, 2009 at 6:38 AM, Rob Charlton<br>
</div>> <<a href="mailto:rob.charlton@savageminds.com">rob.charlton@savageminds.com</a> <mailto:<a href="mailto:rob.charlton@savageminds.com">rob.charlton@savageminds.com</a>>><br>
<div><div></div><div class="Wj3C7c">> wrote:<br>
><br>
> Hi,<br>
><br>
> I'm using mod_event_socket to listen for DTMF events. I have Nokia<br>
> handsets registered as SIP clients over Wifi, as well as a SIP trunk<br>
> providing incoming PSTN calls to a range of DDIs and outgoing PSTN<br>
> calls.<br>
><br>
> If I make an incoming (PSTN or SIP) call and answer it, I always see<br>
> DTMF events via mod_event_socket.<br>
> If I make an outgoing call direct to a handset using SIP then I<br>
> see DTMF<br>
> events - e.g. originate user/1000 &park()<br>
> If I make an outgoing call via PSTN then I don't see DTMF events e.g.<br>
> originate sofia/gateway/mygateway/myphonenumber &park() or<br>
> &javascript(myscript.js);<br>
><br>
> In the latter case, I am still able to pick up DTMF digits if I use<br>
> javascript session.collectInput() - so it appears as if the DTMF tones<br>
> are being recognised by Freeswitch, but no events sent.<br>
><br>
> What am I doing wrong?<br>
><br>
> Cheers<br>
><br>
> Rob<br>
><br>
> --<br>
> Rob Charlton<br>
> Savage Minds Ltd<br>
><br>
<br>
<br>
_______________________________________________<br>
Freeswitch-dev mailing list<br>
<a href="mailto:Freeswitch-dev@lists.freeswitch.org">Freeswitch-dev@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-dev" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-dev</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
<br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com">MSN:anthony_minessale@hotmail.com</a><br>GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org">sip:888@conference.freeswitch.org</a><br><a href="http://iax:guest@conference.freeswitch.org/888">iax:guest@conference.freeswitch.org/888</a><br>
<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org">googletalk:conf+888@conference.freeswitch.org</a><br>pstn:213-799-1400<br>