[Freeswitch-users] problems with portaudio

[Mike Jerris]

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.

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20191203/dfa11974/attachment.html>