similar to: How to increase the performance?

Hi! The encoding, decoding and recoding cost cpu time, that's sure. But does this time differs much depending on the used codec? Is - for example - a G729 faster than a GSM codec? Bye! Michael

Hi! I want to connect my asterisk system to the PSTN. Now I'm thinking about using an analog modem as FXO device. Which modems do run? How is the voice quality? (I looked into the wiki, but on <http://www.voip-info.org/tiki-index.php?page=Asterisk%20hardware> no regular modems were mentioned. Or should I use a special FXO-card? I don't want to spend 99$ for such a thing but

I have two phones (111 and 112) on a LAN, and I have on a users site a phone 333. phone 111 uses sip.conf, while 112 uses real-time set-up. 111 can call 333 AND the audio is working 112 can call 333 but audio is just white noise. 333 can call 111 or 112 and audio is working. The phones are identically set-up (just user name = phone number and password are different) sip.conf (for 111 - all

A Likert scale may have produced counts of answers per category. According to theory I may expect equality over the categories. A statistical test shall reveal the actual equality in my sample. When applying a chi square test with increasing number of repetitions (simulate.p.value) over a fixed sample, the p-value decreases dramatically (looks as if converge to zero). (1) Why? (2) (If

A note to all those who are avoiding writing up an AGI becasue it looks two complicated.. I have just written up my first and its awesome.. It makes Asterisk open to all sorts of possibilities.. let your imagination run wild.. I put off writing an AGI script because a) I could not find any docs b) it looked like the only way to do it was perl and I know nothing about perl and c) I am not a

Has anyone seen this: http://www.microsoft.com/windows/Embedded/sak/evaluation/compare/advantage.asp And has anyone from the Samba team posted a response? It seems like some of the information presented there is contrary to what I know about Samba. ERX

This seems like a piece members of this list would find interesting... === There is growing concern over the interaction of VoIP systems with the legacy PSTN, and the transmission of caller identity data--most notably, Caller ID on the PSTN. It is not always possible, or obvious how, to handle Caller ID data when moving to or from VoIP and the PSTN networks. There are even business models

In 2.4 kernels, there was a [bp]fifo that could be implemented with ''tc add'', but in 2.6 kernels, I find no way to create either of these fifos. I am able to increase txqueuelen. What I desire is a larger number of backlogged packetes before drops occur. How can the output buffer size be increased for HTB? -- gypsy

http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for "free" VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of

Hi experts I have a machine which have 5 hard disk I want to use glusterfs to fast my disk my machine's IP is 123.224.178.67 my steps 1)create the volume gluster volume create vol1 123.224.178.67:/buildarea2 123.224.178.67:/buildarea3 123.224.178.67:/buildarea4 123.224.178.67:/buildarea5 123.224.178.67: /buildarea6 2)start the volume gluster volume start vol1 3)mount the volume

Dear R users, I am running lmer() and having memory problem: "reached total allocation of 1535Mb". The lmer() function was successful (no errors or warnings). However when I do "summary(fit.lmer), I got this "Error: cannot allocate vector of size 19.5 Mb). I used memory.size() and got 1880. I used memory.limit(2500) to increase the memory size to 2500Mb, however I still get

Hi all, We are using - a dell sc440(Single dual-core intel xeon 3040, 1.86GHz,1066MHz front size bus 2MB cache) as the asterisk server - dell 400sc(Intel P4) as a SER server - digium isdn card, TE120P at Asterisk server - Bandwidth: 2Mbps/512kbps All SIP Phones are registered to SER server, and SER will route all outgoing calls to Asterisk server. My problem is the sound quality goes down if

Hi CentOS 7.X, sendmail.x86_64 8.14.7-5.el7, cyrus-sasl.x86_64 2.1.26-23.el7 There are conflicting message on how to increase the logging of saslauthd. I know I can do this: /usr/sbin/saslauthd -d -n0 -m /var/run/saslauthd -a pam but that requires a terminal as saslauthd logs the output to STDOUT, this is not what I want. I would like to have it started as a daemon and verbosity of

I want to full understand register coalescing and how to coalesce copies. From what I have seen from "SimpleRegisterCoalescing," "RegAllocLinearSpan," and "RegAllocPBQP" there are three indicators for the copy instruction x = copy y. Assume that the value number of x does not equal the value number of y. For "x = copy y", let the boolean values a,b, and c,

In commit 08f82f2e3d6975b72340dd59f438862e152a15ef we increased the memory size to 800MB (on x86) so that the semodule command would work. However it has been discovered that another SELinux command takes large amounts of RAM (setfiles during the SELinux relabel step). Therefore increase the memory size again, this time to 2000MB. --- v2v/v2v.ml | 2 +- v2v/virt-v2v.pod | 2 +- 2 files

Hi all, I have released SYSLINUX 3.35-pre5 to the Testing directory; this is a release candidate and I intend to release this as 3.35 early next week unless I hear of any showstopper bugs, so please, pretty please, try it out. http://www.kernel.org/pub/linux/utils/boot/syslinux/Testing/ kernel.org has been overloaded lately, so it might take a while to show up. Changes in 3.35: *

Hello , After upgrading the system from CentOS 7.2.1511 to CenOS 7.3.1611 I see that the average processing time has increased from 5-7% to 12-15% (doubled). Not critical but it is not pleasant. Server as KVM with 5 virtual machines. Someone noticed something similar? If so, how to fix that? Thx.

Hello , After upgrading the system from CentOS 7.2.1511 to CenOS 7.3.1611 I see that the average processing time has increased from 5-7% to 12-15% (doubled). Not critical but it is not pleasant. Server as KVM with 5 virtual machines. Someone noticed something similar? If so, how to fix that? Thx.

I'm hoping this is the the right location for this question.... I've been working with some developers that are desiring to use the multiplexing capabilities of OpenSSH, but have hit the default limit of 10. I've seen some discussions of increasing this default limit previously and have noticed that some of the Linux distros have increased the number to 64; I'm curious of the

Hello, As an exercice, I installed sipp on the same box as a Asterisk 11.4 instance (to keep network equipements out of the equation). I'm focusing on the maximum number of new calls this Asterisk instance can deal with. Here is the dialplan (AEL) I'm playing with: _X. => { Verbose(0,Incoming call from ${CALLERID(num)} to ${EXTEN} in ${CONTEXT} - case A);