Rupa Schomaker wrote:
> Sound bugish to me - or at least not desired behavior.
>
> I'd suggest opening up a jira (jira.freeswitch.org) with as much
> documentation as you have so it can be researched and resolved.
If someone could add it to Jira, I'll detail the issue here. The Jira Web
interfac
Also, in general, I believe you want the jitter buffers on the end-point
devices only. Not the guy in the middle. So, jitter buffer should be
enabled on the phone, not within FS -- unless FS is the endpoint (eg: IVR).
On Fri, May 8, 2009 at 7:05 AM, Rupa Schomaker wrote:
> Sound bugish to me -
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much
documentation as you have so it can be researched and resolved.
On Fri, May 8, 2009 at 3:46 AM, Jason White wrote:
> Sorry for all the e-mail...
>
> If I turn off the jitter b
Sorry for all the e-mail...
If I turn off the jitter buffer that I had set in the dialplan extension for
that provider, DTMF is correctly sent and detected by the other side.
I suspect a bug, but maybe this is the desired behaviour.
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As a matter of interest, the other end (as reported in its SDP) is BroadWorks.
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Jason White wrote:
>It is also called if I use the voicemail
> extension on my local FreeSWITCH.
Apologies for the nonsense - I meant that switch_rtp_dequeue_dtmf() is called
in that case, for DTMF detection.
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I've narrowed this problem down.
When I call my ISP's DTMF test and issue DTMF from the Snom phone, do_2833()
from switch_rtp.c is never called, as evidenced by freeswitch.log.
However, if I call a friend's FreeSWITCH box from the phone (via my FreeSWITCH
instance), do_2833() is called. It is als
Anthony Minessale wrote:
> you may have a sonus infection
>
> try some of the stuff from here under DTMF
>
> http://wiki.freeswitch.org/wiki/RTP_Issues
Thank you for the suggestion.
I tried both the Sonus and Cisco settings in the external profile (running
sofia profile external restart reload
you may have a sonus infection
try some of the stuff from here under DTMF
http://wiki.freeswitch.org/wiki/RTP_Issues
On Thu, May 7, 2009 at 5:16 AM, Jason White wrote:
> Remko Kloosterman wrote:
> >
> >
> > Did you make a wireshark trace yet? You should be able to find out
> > exactly what's
Remko Kloosterman wrote:
>
>
> Did you make a wireshark trace yet? You should be able to find out
> exactly what's going on there, which protocol is used, etc. We've had
> our share of problems with DTMF over SIP trunks as well.
I've just discovered that I'm having a similar problem to the on
g] Namens Jay Austad
Verzonden: woensdag 6 mei 2009 20:57
Aan: freeswitch-users@lists.freeswitch.org
Onderwerp: Re: [Freeswitch-users] DTMF recognition flaky
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed any
009 19:57
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] DTMF recognition flaky
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed anything. I'm using
voicepulse for my SIP trunks. Is the
I'm running 1.0.4pre3. Haven't gotten a chance to upgrade to pre7 yet.
2833 is the default right? I haven't changed anything. I'm using
voicepulse for my SIP trunks. Is there an option I can add to that
definition to force RFC2833?
--
jay austad | 612.423.1433 | aus...@signal15.com
Well it depends.. first off are you doing inband dtmf or RFC2833?
Secondly what SVN rev are you running?
/b
On May 6, 2009, at 1:44 PM, Jay Austad wrote:
Using the default installation, I've noticed that when I (or someone
else) calls in on my SIP trunk and keys in an extension, not all of
Using the default installation, I've noticed that when I (or someone
else) calls in on my SIP trunk and keys in an extension, not all of
the numbers are recognized unless they hold the key down for at least
1/2 second to a second.
Is there a way to improve DTMF recognition so people can just
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