similar to: Occasional robotic sound while call in progress

I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present. Can someone break down that dial string with an explanation? The 011 look like an overseas call (from Americas), while the 972595XXXXXX is unclear... -------------- next part -------------- An HTML attachment was scrubbed... URL:

We've deoplyed a number of pure VoIP wireless (wifi & proprietary) phones, but not dect. Is there a simple overview of integrating DECT phones with Asterisk somewhere? I assume the DECT basestation has a multi-account SIP VoIP interface, and the handsets are just plain old dect? Can you push configuration info to individual phones? (Are they individually addressible / configurable

Is there a simple way to get call volume from a particular trunk within the dialplan (for conditional branching)? I suspect we will have to build an AGI script but I'm hoping something new in Asterisk 13 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150627/6774c750/attachment.html>

Is there a recent survey of that Linux distro and version people are using for the Asterisk installations? I recall seeing a pie chart over a year ago (I think on a wiki but I can't find it again)....also hoping for something more current. I suspect RH5 and RH6 are most popular...but I'm looking for facts -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD

I've searched through the wiki but I can't find what I need...I'm trying to figure out what the max call duation is. I found references to "show application AbsoluteTimeout" but that isn't in 1.6 (not even prepending "core" to the front). A core help show didn't help... -------------- next part -------------- An HTML attachment was scrubbed... URL:

I happened to be in the cli tonight as some (208.122.57.58) initiated a simple attack - just trying to make long distance calls from outside context. Although harmless, this went on for several minutes as the idiot just used up my bandwidth with SIP messages. Here's and example: [2011-12-28 22:53:42] NOTICE[9635]: chan_sip.c:14035 handle_request_invite: Call from '' to extension

Is the a CLI command that shows all channels in use at one time? (Whether IAX, SIP, SCCP, etc)? As well, when I "SIP SHOW CHANNELS" I see phones registering showing as channels in use. Is there a way to filter this output? Thanks! MD

I need to disable/enable a peer after hours automatically, and am thinking about doing so via the AMI. Is there a command to enable/disable (or perhaps delete/add) a peer via the AMI? I could create code to modify sip.conf and force a reload, but that seems like the wrong approach... -------------- next part -------------- An HTML attachment was scrubbed... URL:

Can someone tell me when the /proc/dahdi files are created for spans? Are they created when asterisk starts (or the asterisk init script) - if not what script creates them? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150204/f14ae5a2/attachment.html>

I have a custom voicemail script which reformats and forwards the attached voicemail wav file to the recipient. I would like to make use of a channel variable in my script; is there a way to pass a channel variable to this voicemail script? -------------- next part -------------- An HTML attachment was scrubbed... URL:

After reading about the 2 major SSL (and TLS?) weaknesses discovered this year, I was wondering how it affects asterisk. Does the SIP authentication use TLS - or something that was recently broken? Is there a risk of exposing passwords? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

We have a small office installation running over a cable modem. (8M down, 500k up confirmed with numerous speed test sites) When a single call is up, call quality is fine. When a second call is up, outbound audio is immediately choppy. We're using ulaw, and confirmed that traffic with 2 calls is <175kbps in/out. (IAX connection out) Asterisk doesn't report any dropped frames, the

?I meant how many calls are in progress on a particular trunk. (Sorry - I didn't even think of the other interpretation). ________________________________ From: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at lists.digium.com> on behalf of Matt Riddell <lists at venturevoip.com> Sent: Sunday, June 28, 2015 9:26 AM To: Asterisk Users List Subject: Re:

On a Compaq DL360 G1 w/ dual PIII 1.27's I installed Centos 5 minimal and then a yum update When I do a "uname -a" I get this Linux tstsrvr.abbacomm.net 2.6.18-8.1.3.el5 #1 SMP Mon Apr 30 19:55:44 EDT 2007 i686 i686 i386 GNU/Linux When I do a rpm -qa | grep kern* I get this kernel-devel-2.6.18-8.1.3.el5 kernel-2.6.18-8.1.3.el5 kernel-headers-2.6.18-8.1.3.el5 It is interesting

I am specifically using Celt Wideband (48kHz) over WiFi multicast that naturally leads to lost packets and am trying to minimize the impact to the audio. I implemented PLC but the audio it produces is robotic. Have I implemented PLC correctly? Checking the waveform it is using the previous received waveform to fill in a missing packet but not the full waveform so it has to repeat. Basically,

We're looking at a large wifi phone deployment, and we're looking for wifi phones that: 1. Are SIP compliant (Asterisk friendly) 2. Provision capable (ideally TFTP of own MAC address) 3. Industrial quality (no cheap plastic stuff). 4. Well documented (and none of the "only telco's get documentation" crap) Does anyone have a suggestion? Thanks, MD -------------- next

Hi, I'm currently working on the TI DaVinci processor, which has an ARM926 processor and a TMS320C64+ DSP in it. I have been using Speex 6kbps (narrowband, quality=2, complexity=1, perceptual_enh=0) on the DSP C64+ side without any issues; the voice sounds very good for such a low bandwidth requirement. However, recently, I tried out the 4kbps mode (changing quality to 1) on the DSP and it

Tried your fixed_generic.h change but that didn't help. Andy ----- Original Message ---- From: Andy Ngo <ndno72-speex@yahoo.com> To: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Sent: Wednesday, December 12, 2007 4:13:35 PM Subject: Re: [Speex-dev] 4kbps sounds robotic on TMS320C64 Jean-Marc, Yes, fixed-point is enabled (#define FIXED_POINT in