[Freeswitch-users] problems with portaudio

[John Covici]

the call is *398300 so  you can find it.

On Wed, 04 Dec 2019 00:14:02 -0500,
Mike Jerris wrote:
> 
> [1  <multipart/alternative (7bit)>]
> [1.1  <text/plain; us-ascii (quoted-printable)>]
> Needs the sip in the pcap too to understand what the rtp is. this looks like capture started after call was already going.
> 
> > On Dec 3, 2019, at 7:39 PM, John Covici <covici at ccs.covici.com> wrote:
> > 
> > Thanks  for your response.
> > 
> > I don't know how to read the pcap file, so here is  a link to the
> > file, maybe you can figure it out better than I can.
> > 
> > https://covici.com/owncloud/index.php/s/yHS2Tc4Z2FEeGZb <https://covici.com/owncloud/index.php/s/yHS2Tc4Z2FEeGZb>
> > 
> > 
> > On Tue, 03 Dec 2019 18:52:41 -0500,
> > Mike Jerris wrote:
> >> 
> >> [1  <multipart/alternative (7bit)>]
> >> [1.1  <text/plain; us-ascii (quoted-printable)>]
> >> Can you check a pcap to confirm.  This MAY be an issue I just saw last week and have to do with rtp timestamps.  If you can confirm that rtp is sending out but the timestamps dont seem right, that would confirm it.  
> >> 
> >>> On Nov 29, 2019, at 11:28 AM, John Covici <covici at ccs.covici.com> wrote:
> >>> 
> >>> I have the log of the call which looks normal.  My guess is that  rtp
> >>> is not properly being sent out, for some reason.  The hangup cause is
> >>> always normal_clearing.
> >>> 
> >>> On Fri, 29 Nov 2019 13:06:12 -0500,
> >>> David Villasmil wrote:
> >>>> 
> >>>> [1  <multipart/alternative (7bit)>]
> >>>> [1.1  <text/plain; UTF-8 (7bit)>]
> >>>> Do you have any trace?
> >>>> 
> >>>> On Fri, 29 Nov 2019 at 18:05, John Covici <covici at ccs.covici.com> wrote:
> >>>> 
> >>>>> Some more information -- even after pressing a digit and getting
> >>>>> audio, it hangs up after about 30 seconds.
> >>>>> 
> >>>>> On Fri, 29 Nov 2019 10:46:21 -0500,
> >>>>> John Covici wrote:
> >>>>>> 
> >>>>>> Hi.  I finally was able to upgrade fs to master as of llast night.
> >>>>>> Its working well, except if I use portaudio to make a call.  This all
> >>>>>> worked find in fs 1.6.20.
> >>>>>> 
> >>>>>> When I call someone I cannot hear anything until I send it a dtmf
> >>>>>> (rfc2283) and then things work normally, at least I can hear
> >>>>>> something.  I had a look at the logs, but nothing strange in there
> >>>>>> after typing the digit.
> >>>>>> 
> >>>>>> Also, I cannot call a local extension from port audio, even though the
> >>>>>> extension is registered and can be called from another extension.  It
> >>>>>> immediately goes to voicemail.
> >>>>>> 
> >>>>>> Thanks in advance for any suggestions.
> 
> [1.2  <text/html; us-ascii (quoted-printable)>]
> [2  <text/plain; utf-8 (base64)>]
> _________________________________________________________________________
> 
> The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
> Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
> Build your next product on our scalable cloud platform.
> 
> Join our online community to chat in real time https://signalwire.community
> 
> Professional FreeSWITCH Services
> sales at freeswitch.com
> https://freeswitch.com
> 
> Official FreeSWITCH Sites
> https://freeswitch.com/oss
> https://freeswitch.org/confluence
> https://cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici wb2una
         covici at ccs.covici.com