All I have to do to make it work is to use 1.8.0 revision 281875 --
after that something is broke. I was hoping someone could look and see
what changed just after that rev and see if it makes sense.
Benny Amorsen wrote:
> cov...@ccs.covici.com writes:
>
> > But it surpresses in both directions
cov...@ccs.covici.com writes:
> But it surpresses in both directions! I still want to hear the other
> end. For a test is there a way to turn off that feature to see if that
> is the cause?
Ah, so it isn't Asterisk doing silence suppression, it's Asterisk being
unable to handle that other devic
Philipp von Klitzing wrote:
> Hi!
>
> >> Why is it a problem? It sounds like Asterisk does silence suppression.
> >
> > 1) With no rtp traffic, the nat device will drop the connection in it's
> > nat table and thus disconnecting the softphone from Asterisk. (after
> > the router's timeout peri
Hi!
>> Why is it a problem? It sounds like Asterisk does silence suppression.
>
> 1) With no rtp traffic, the nat device will drop the connection in it's
> nat table and thus disconnecting the softphone from Asterisk. (after
> the router's timeout period of course)
>
> 2) The other issue is you
Lyle Giese wrote:
> Benny Amorsen wrote:
> > cov...@ccs.covici.com writes:
> >
> >
> >> Hi. I am having a very strange problem --aren't they all -- with the
> >> release candidate. I have softphone which talks to asterisk from behind
> >> nat -- the asterisk is on a public ip -- and when I h
Benny Amorsen wrote:
> cov...@ccs.covici.com writes:
>
>
>> Hi. I am having a very strange problem --aren't they all -- with the
>> release candidate. I have softphone which talks to asterisk from behind
>> nat -- the asterisk is on a public ip -- and when I hit mute on the
>> softphone, all r
Benny Amorsen wrote:
> cov...@ccs.covici.com writes:
>
> > Hi. I am having a very strange problem --aren't they all -- with the
> > release candidate. I have softphone which talks to asterisk from behind
> > nat -- the asterisk is on a public ip -- and when I hit mute on the
> > softphone, all
Leif Madsen wrote:
> On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote:
> > Hi. I am having a very strange problem --aren't they all -- with the
> > release candidate. I have softphone which talks to asterisk from behind
> > nat -- the asterisk is on a public ip -- and when I hit mute on the
>
cov...@ccs.covici.com writes:
> Hi. I am having a very strange problem --aren't they all -- with the
> release candidate. I have softphone which talks to asterisk from behind
> nat -- the asterisk is on a public ip -- and when I hit mute on the
> softphone, all rtp traffic ceases! Now, a versio
On 10-09-23 05:40 PM, cov...@ccs.covici.com wrote:
> Hi. I am having a very strange problem --aren't they all -- with the
> release candidate. I have softphone which talks to asterisk from behind
> nat -- the asterisk is on a public ip -- and when I hit mute on the
> softphone, all rtp traffic ce
Hi. I am having a very strange problem --aren't they all -- with the
release candidate. I have softphone which talks to asterisk from behind
nat -- the asterisk is on a public ip -- and when I hit mute on the
softphone, all rtp traffic ceases! Now, a version which does work is
r281875, this does
On Thu, 2005-04-14 at 07:23 -0500, Eric Wieling wrote:
> Is Asterisk getting a stream of RTP packets from the SIP client? What
> happens if you start talking on the SIP device? Does Asterisk then
> start sending RTP? It still sounds like VAD and silence supression is
> enabled on the SIP dev
trixter http://www.0xdecafbad.com wrote:
I have done some further research, the first RTP packet is sent when
playback() is called. No others. The application is running, if I
press a key and goto a different item that would cause a new
playback()/background() 1 more RTP packet is sent.
To be
I have done some further research, the first RTP packet is sent when
playback() is called. No others. The application is running, if I
press a key and goto a different item that would cause a new
playback()/background() 1 more RTP packet is sent.
To be clear If I call myself, RTP packets are s
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