Displaying 20 results from an estimated 320 matches for "ptimes".
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times
2019 Sep 03
2
ptime
We have a customer with a system rejecting calls from Asterisk. It's indicating the ptime is 60, but the system admin is saying they only support 20.
They are running asterisk 16.2.1 and using chan_sip
Is there a way to specify this with chan_sip?
Also, for my own curiosity, is there a way to specify this with PJSIP? (Trying to migrate customers to PJSIP, but we are holding until asterisk
2009 Mar 14
2
Format about Date and time
I have a data set like this:
> head(FormatedData)
ID Target Actual Date Time
1 2030 0 -57.0 12/20/08 17:03:00
2 2030 90 90.0 12/20/08 18:41:00
3 2030 45 43.8 12/21/08 14:36:00
4 2030 0 -23.8 12/21/08 19:30:00
5 2030 90 90.2 12/21/08 21:48:00
6 2030 45 48.6 12/22/08 13:02:00
I wan to convert the format of Date and Time, so I did this:
pdate
2013 Jan 30
1
R CMD check: Error in get("ptime", pos = "CheckExEnv") ...
Hi,
Does anyboody have insight into what this error terminating "R CMD check" on
an in-house package may imply?
> ###
> cat("Time elapsed: ", proc.time() - get("ptime", pos = 'CheckExEnv'),"\n")
Error in get("ptime", pos = "CheckExEnv") :
unused argument(s) (pos = "CheckExEnv")
Calls: cat -> cat.default
2008 Jan 22
3
gctorture and proc.time (PR#10600)
In R version 2.6.1 (2007-11-26)
and R version 2.6.1 Patched (2008-01-19 r44061)
on openSUSE 10.2 (X86-64)
> gctorture()
> proc.time()
Error: protect(): protection stack overflow
The problem with this is that then
R CMD check --use-gct foo
ALWAYS FAILS with
> cat("Time elapsed: ", proc.time() - get("ptime", pos = 'CheckExEnv'),"\n")
Error in
2007 May 16
2
draft-ietf-avt-rtp-speex-01.txt
> Page 3:
>
> To be compliant with this specification, implementations MUST support
> 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate.
> The sampling rate MUST be 8, 16 or 32 kHz.
>
> There is a type above after (narrowband), there is a " extra character.
>
> I don't understand what is the motivation to specify "SHOULD
2006 Jan 13
26
A couple of issues
I''ve been testing ZFS since it came out on b27 and this week I BFUed to b30. I''ve seen two problems, one I''ll call minor and the other major. The hardware is a Dell PowerEdge 2600 with 2 3.2GHz Xeons, 2GB memory and a perc3 controller. I have created a filesystem for over 1000 users on it and take hourly snapshots, which destroy the one from 24 hours ago, except the
2006 Jun 12
3
compare Date with TIme
Hi
When returning data from a database column set as a date field I get
''2006-06-06''
I am then have these two lines of code in my controller, taht gain the
date I require.
pNow = Time.now
@pDate = Time.local(pNow.year, pNow.month, 1)
At the monment one is set to a date data type while one is set to a Time
data type.
how can i change these to be the same data type so i can
2006 Oct 12
0
Codes negotiation problemsbetweenAsterisk1.4beta2 and Aastra 480i
The problem with the extra ptime descriptions in the SDP has been fixed in Asterisk (see http://lists.digium.com/pipermail/svn-commits/2006-October/017694.html). I've got the latest version of the 1.4 branch from SVN and have verified that the codec negotiation is working again.
If you don't want to try the latest SVN version, then you'll have to restrict the phones to a single codec
2007 May 15
4
draft-ietf-avt-rtp-speex-01.txt
Hi all
We are about to send an updated version of the internet draft
"RTP Payload Format for the Speex Codec" to the IETF AVT working group.
Before submitting we would like your input, if you have any comments
or input please send them to the mailing list.
If we don't get any comments in 1 week (by 22. May 2007) we will go ahead
and submit it to the IETF. Of course you can comment
2006 Mar 08
1
power and sample size for a GLM with Poisson response variable
Craig, Thanks for your follow-up note on using the asypow package. My
problem was not only constructing the "constraints" vector but, for my
particular situation (Poisson regression, two groups, sample sizes of
(1081,3180), I get very different results using asypow package compared
to my other (home grown) approaches.
library(asypow)
pois.mean<-c(0.0065,0.0003)
info.pois <-
2007 May 16
0
draft-ietf-avt-rtp-speex-01.txt
comment inline.
On Wed, 16 May 2007, Jean-Marc Valin wrote:
>> Page 3:
>>
>> To be compliant with this specification, implementations MUST support
>> 8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate.
>> The sampling rate MUST be 8, 16 or 32 kHz.
>>
>> There is a type above after (narrowband), there is a " extra
2007 May 15
0
draft-ietf-avt-rtp-speex-01.txt
Here my comments:
Page 3:
To be compliant with this specification, implementations MUST support
8 kHz sampling rate (narrowband)" and SHOULD support 8 kbps bitrate.
The sampling rate MUST be 8, 16 or 32 kHz.
There is a type above after (narrowband), there is a " extra character.
I don't understand what is the motivation to specify "SHOULD support 8
kbps
2009 Feb 13
3
Strange performance loss
I''m moving some data off an old machine to something reasonably new.
Normally, the new machine performs better, but I have one case just now
where the new system is terribly slow.
Old machine - V880 (Solaris 8) with SVM raid-5:
# ptime du -kds foo
15043722 foo
real 6.955
user 0.964
sys 5.492
And now the new machine - T5140 (latest Solaris 10) with ZFS
2008 Oct 27
1
Forcing repacketization on SIP to SIP call
Hi folks
I have a handset talking to Asterisk, which in turn puts the call through to
an ITSP.
The handsets likes to send audio in 40ms frames (even though Asterisk
requests 20ms frames with a ptime header in the SDP).
The ITSP doesn't request any particular frame length (with ptime) or state a
maximum length (with maxptime), so when Asterisk receives the 40ms media
frames from the handset,
2013 Mar 09
0
About Zitter Control
Hello there,
I have built a kernel module that is responsible for sip and rtp
encryption/decryption, padding/depadding, ptime decrease (splitting a
large rtp packet to splitting into smaller packets of ptime 20) for
incoming packets and ptime increase (merging small rtp packets into a
big rtp packet of desired ptime) for outgoing packets. This is meant
to be a faster process and indeed it is. The
2013 Sep 17
1
RTP not being switched between both SIP endpoints
We have a system where calls are coming in from telcos via an opensips
server and then being redirected out to a regular sip destination.
There is no NAT, DTMF features, call recording, or codec translation
being performed so I would expect asterisk to issue a reinvite after the
call is answered and switch the audio however it is not happening.
Here is the sip peer information for the call
2006 Apr 05
2
Setting ptime attribute in SDP invite
Is it possible for Asterisk to set the ptime attribute on outbound calls in
SDP invite?
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2006 Feb 06
3
power and sample size for a GLM with poisson response variable
Hi all,
I would like to estimate power and necessary sample size for a GLM with
a response variable that has a poisson distribution. Do you have any
suggestions for how I can do this in R? Thank you for your help.
Sincerely,
Craig
--
Craig A. Faulhaber
Department of Forest, Range, and Wildlife Sciences
Utah State University
5230 Old Main Hill
Logan, UT 84322
(435)797-3892
2016 Oct 15
2
Registered successfully, but after a minute or so no SIP messages anymore
hi,
let me explain in detail, what i have configured and what is happening now:
1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to
device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device
192.168.3.99 (same ports)
other direction is totally open.
I
2007 Jun 07
1
draft-ietf-avt-rtp-speex-01.txt
Looks good to me.
Jean-Marc
Alfred E. Heggestad a ?crit :
> Hi
>
> Please find an updated version of the Speex I-D attached. The only
> change is addition of the copyright conditions in Appendix A,
> as requested by Ivo.
>
> Many thanks for your input.
>
> I will give you a few more days before submitting to AVT working group
>
>
> /alfred
>
> Ivo