27 feb 2010 kl. 08.26 skrev Olle E. Johansson:
26 feb 2010 kl. 22.02 skrev JT:
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat
of a band-aid to the issue. But in my observations there is one clear
indicator that I am shocked is not used.
When I have done
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat
of a band-aid to the issue. But in my observations there is one clear
indicator that I am shocked is not used.
When I have done this test - pulling the network cable on a device during a
call - Asterisk actually reports
26 feb 2010 kl. 22.02 skrev JT:
Hmmm I agree that altering sip.conf with the RTP timeouts are somewhat of
a band-aid to the issue. But in my observations there is one clear indicator
that I am shocked is not used.
When I have done this test - pulling the network cable on a device
Hello,
worst aspect is that - if SIP clients do not have such a timeout, and in
that case if killing an asterisk and to start it up again -
so it is nothing to do with this asterisk timeout.
Regards,
On 23 February 2010 08:44, Olle E. Johansson o...@edvina.net wrote:
23 feb 2010 kl. 01.47
Good day all!
I have an issue which has plagued me for quite sometime now...and as I close
in on its cause, I have reached a point where additional info would be
greatly helpful!
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one
On Mon, 2010-02-22 at 16:13 -0500, JT wrote:
Is this something that is fixed in an update? (Currently running 1.2)
Yes... modern versions of Asterisk support SIP session timers. (If I
remember correctly, Asterisk 1.2 could tear down a call based on lack of
RTP data, but I never found it worked
On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one of those SIP devices hangs up, Asterisk receives the hangup notice
and disconnects the call/channel.
However - what does Asterisk do when the network
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the client is
going away, as no activity is a good
On 100222 1818, Kevin P. Fleming wrote:
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the client is
23 feb 2010 kl. 03.18 skrev Kevin P. Fleming:
Kirill 'Big K' Katsnelson wrote:
The caveat here is that it is perfectly normal NOT to transmit any RTP
data in case of long silence. This is why the SIP timers were introduced
in the first place: there is no correct way to detect when the
23 feb 2010 kl. 01.47 skrev Kirill 'Big K' Katsnelson:
On 100222 1313, JT wrote:
When a SIP device dials another SIP device...Asterisk connects the calls and
displays the channel information.
If one of those SIP devices hangs up, Asterisk receives the hangup notice
and disconnects the
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