OK, I have updated the file including portrange 5060-5080 as well as the rtp portrange. The same link should still work. Thanks. 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@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@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@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.
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