similar to: OT: script to remove leading and trailing silence

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

On Thu, 17 Apr 2014, Jerry Geis wrote: > I was thinking transcoding was through PRI card - not gsm to ulaw. :) You can convert the GSM files to ULAW using sox. I tend to transcode everything to WAV (PCM not that funky 'GSM in WAV') because it is relatively cheap (CPU cycles) to transcode from WAV to ULAW and everything else in the world understands WAV just fine. If you really need

On Sat, Aug 29, 2009 at 10:30 PM, <asterisk-users-request at lists.digium.com>wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help'

Zonbu.com sells a little box I was planning to use as a router, but I couldn't resist putting Asterisk on it just for fun. It may never see its intended purpose. The box costs US$249 (and was delivered 40 hours after being orderd!), but you can get it for less if you subscribe to their service. I didn't. The box is an Intel compatible processor (VIA Esther processor 1200MHz) with

I'm trying to determine the capacity of my host running Asterisk 11.8.1 on CentOS 6.5. The host is an Intel E3-1240v3 with 8GB RAM, an SSD, and gigabit Ethernet. The primary application will be bridging groups of users using meetme(). I'm using 2 boxes -- 1 to initiate calls using call files (box1), and 1 behaving a bit more like a production box -- bridging calls (box2). The call

is typing his number, though there is a 15 seconds timeout, and even if = I type the number very fast it still may happen to me.<o:p></o:p></p><p = class=3DMsoNormal><o:p>&nbsp;</o:p></p></div><p class=3DMsoNormal>It has = been my casual observation that the speed at which I enter digits on my = phone is unrelated to the speed at which my

I was debugging a turnup with Global Crossing the other day and they presented me with a web page that displayed a 'ladder diagram' of a call including a ton of detail all neatly organized in tabs and links so you could drill down to any level of detail needed. The copyright notice says 'Copyright? 2008 Empirix.' Is there any free software available to analyze a pcap or

I have Sipura 841's talking to a CVS-HEAD of August 4. If I disconnect the power to the Sipura, Asterisk does not hang up the channel. My sip.conf for this phone looks like: ; [super1] ; Sipura 841 disallow = all allow = ulaw callerid = "super1"

"asterisk-users-request at lists.digium.com" <asterisk-users-request at lists.digium.com> wrote: Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to

I'm starting a new project similar to a previous project where I used OpenSER to front a bunch of Asterisk servers. Now that OpenSER is gone, OpenSIPS and Kamailio seem the likely candidates. I'm leaning towards OpenSIPS because it's in EPEL so I can install it with yum. Also, because I think the name sounds more 'professional' when discussing architecture with clients :)

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

On Mon, 27 Apr 2015, Bryant Zimmerman wrote: > exten => _9XXXXXXX,n,Set(dialnumber=${l_HomeAreaCode}${EXTEN-1}) Missing a colon? ${EXTEN:-1} -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

Does anybody have contact info for Dan Toma, the author of Diax? I've tried danto at clicknet.ro and danto at rdslink.ro without success. Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000

A client asked me to come with a system that will pass certification with PPP in the UK. Google is not being helpful :( It has something to do with recording calls in case the PPP requests a copy. Supposedly their rules were relaxed on August 1st if that helps. Any clues (links?) will be appreciated. -- Thanks in advance,

On Wed, 25 Feb 2015, A J Stiles wrote: > The limiting factor with a switch carrying IP telephony traffic is not > bandwidth, but routing table entries; and even cheap switches nowadays > will usually take 1024 entries, if not 4096. Are you referring to the MAC CAM table? Saying 'routing table' and 'switch' in the same sentence seems confusing. Do VOIP devices take

I've googled for way too long, where are the 2 letter language values defined? I know: en = English es = Spanish fr = French but what about Croatian, Russian, Serbian, Vulcan, etc? Is there a list documented for Asterisk or is it "just use the 2 letter country code Internet TLD?" Thanks in advance, ------------------------------------------------------------------------ Steve

On Sun, 31 May 2015, Luca Bertoncello wrote: > Now, it would be nice, if I can signaling on the phone which number will > be called, so that, for example, if I receive a call for +493511111111 I > get a message on the display or the phone ring with a particular tone, > and if I receive a call for +493512222222 the phone write something > other on the display or ring with

On Fri, 26 Jun 2015, Dale Noll wrote: > I added a filter to the /etc/rsyslog.conf file > > :syslogtag, contains, "asterisk" stop > > Syslog is still receiving the messages, but is discarding them. Nice to learn a new (to me) feature of rsyslog. What does 'logger show channels' show? -- Thanks in advance,

According to http://kb.digium.com/entry/12/ The Iaxy will respond to pings on port 9999. You can ping your broadcast IP on your network and listen with tcpdump on your network on port 9999 which will show the Iaxy responding and what IP address it is coming from. Ex. ping 192.168.1.255 tcpdump -i eth0 udp port 9999" Before I get my karma whacked again, does this work for

I'm doing a new install for an old customer. The customer is running a custom version of Asterisk based on version 1.2.7.1. It works for them -- aside from a memory leak requiring a restart once every couple of months... I think the "corresponding" version of Zaptel is 1.2.5, but I'd like to run a bit more modern like Zaptel 1.2.27. Am I just asking for trouble? Thanks in